Re: [Asterisk-Users] * Server behind a firewall - How To
> My * server is NAT'd behind a firewall. > What ports do I need to open to allow a Grandstream IP to connect to it > remotely? You should read the wiki pages given above, but here is what I've done on my linksys: 4569 --> * 5060 --> * 1-10100 --> * in rtp.conf rtpstart=1 rtpend=10100 in sip.conf externip=123.123.123.123 I think that's all I had to do. -- When a simple answer can be given, it makes searching the list easier. ___ 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] IaXY MWI
On Sat, 16 Oct 2004 06:54:16 -0400, Kevin <[EMAIL PROTECTED]> wrote: > I tried the mailbox statement and it didn't work. Include the voicemail.conf context as in voicemail.conf [users] 1234 => ,Joe Blow,[EMAIL PROTECTED],, ; etc [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bandwidth control on a home office network
Looks like it's time to add a WIKI page on QOS routing alternatives, listing options such as the Linksys WRT (with OpenWRT or Sveasoft or...), m0n0wall, LEAF, etc. It seems that this would be a bit off-topic, but QOS if very much a concern for VOIP. Any volunteers who'd actually know what they're talking about? I'm currently in the research phase of my next router-solution, since it's good-bye for my trusted 5861 soon. > -Original Message- > From: Roger Hanson [mailto:[EMAIL PROTECTED] > Sent: Sunday, October 17, 2004 1:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Bandwidth control on a home > office network > > > > - Original Message - > From: "Adam Holt" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > > > > > Hi, > > > > I have a Grandstream ATA today connected to my 750k broadband > > connection via an older router / firewall that doesn't have > any QoS / > > ToS capability. It works fine apart from the obvious > problem of when > > large emails come in or somebody else on the network starts d/l-ing > > something big off the web. > > > > I'm wondering whether to swap the router for a Cisco in order to > > introduce some local bandwidth control. > > > > Alternatively I was wondering if I picked up a Cisco 7960 handset > > instead - is the 2nd ethernet port routed through the > device, or does > > it just act as an Ethernet repeater, i.e. if I arranged the > handset in > > the network as below would I get bandwidth prioritisation for the > > 7960? > > > > [CABLE MODEM]--[7960]---[FW / ROUTER / > HUB][REST OF MY > > NETWORK] > > > > Thanks for any tips. > > > > BR /adam. > > > > I use IPCOP - it's another open source project. It does traffic > shaping, routing, firewalling, DMZ, etc. It's free and runs > on an old > PC (I use Pentium 200MHZ w/128MB RAM - but I need it that > fast because > it's also a content filter for my home network/kids. > www.ipcop.org Did I mention, it's free? Roger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwidth control on a home office network
Jay Milk wrote: > Looks like it's time to add a WIKI page on QOS routing alternatives, > listing options such as the Linksys WRT (with OpenWRT or Sveasoft > or...), m0n0wall, LEAF, etc. It seems that this would be a bit > off-topic, but QOS if very much a concern for VOIP. Any volunteers > who'd actually know what they're talking about? I'm currently in the > research phase of my next router-solution, since it's good-bye for my > trusted 5861 soon. Qos in general http://www.voip-info.org/tiki-index.php?page=QoS Small/Home Routers with QoS http://www.voip-info.org/wiki-VOIP+Routers Please add information! Thanks. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem building Zaptel from CVS v1-0
When trying to build the zaptel source from the v1-0 label, I get an error compiling the new file wcte11xp.c: parse error before "spinlock_t" This error doesn't occur when compiling the zaptel source from HEAD nor did it occur in the source from Sept 19th. Can anyone suggest the right course of action to take to be able to work with a stable version? For example v1-0-1 compiles just fine (doesn't include the new file) is this an OK stable version to use? I ask because the advice on the Asterisk download pages is to use v1-0. Thanks Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: cisco ip 7905 legal ..
Hi Pavel Do you know where I could buy one of these phones from? I'm based in the UK, but would like to play around with one. I emailed Intracom but got no response. On Thu, 14 Oct 2004 22:59:33 +0200, Pavel Jezek <[EMAIL PROTECTED]> wrote: > maybe Polycom is good phone, but Netphone seems to be cheaper (104, 114, 124 euro), > and (maybe) with more features: > > in-line power (standard 802.3af & cisco poe) > integrated switch (voice VLAN capable, learn via CDP!) > XML browser (callmanager compatible) > corporate phonebook! > SIP, h323, sccp! > speaker phone / handsfree (like cisco 7940/60G) > dual-line (like 7940G) > 128x64 pixel display > codecs: 711/729 (726,723) > SMS support, MP3 streaming :-) > power consumption: 4W (netphone) vs. 6,3W (cisco) > > http://netphone.intracom.gr/english.htm > we plan to buy some units to lab , so I'm looking forward to testing:-) > PJ > > > > > - Original Message - > From: Eric Wieling > Newsgroups: gmane.comp.telephony.pbx.asterisk.user > Sent: Wednesday, October 13, 2004 3:12 PM > Subject: Re: Re: cisco ip 7905 legal .. > > > > Mine is the Polycom Soundpoint IP 500. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automated calling/Bridging and takedown in Asterisk?
I am working on a web phone interface to give normal phonesets more 'virtual buttons'..etc, like the expensive executive phones via control via the web. This lead me to the following issue: I am wondering if it is possible (it doesn't seem to as far as I can tell) to make a script (AGI or otherwise) that will have asterisk automatically do the following without the user needing to originate any calls on their telephone: -Call an extension, hold on to the call (call A) -Call another extension, hold on to the call (call B) -Bridge the two calls (A and B) together (so the two extensions can talk to each other) -Later the script drops call B, but keeps call A up -Then asterisk calls another extension (call C) -Then asterisk bridges A with C so then they can talk to each other ..then later the same thing again (call D, then bridge with A)..etc...etc, Allthis would be AGI or script driven without any user having to press anything on his phone. Is this possible (the main issue I see in asterisk is that I cannot find a command in the asterisk API to bridge/unbridge calls like this without something being originated by a call into asterisk from a user). I looked at meetme, but it doesn't seem appropriate for what I want to do above. Any ideas? Thank you! Jack ___ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com ___ 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] Automated calling/Bridging and takedown in Asterisk?
Jack, I'd suggest using the .call files to initiate your call to phone a - the call script will automatically bridge the call to a destination station. From that you could simply create a goto loop in extenstions.conf that calls the same agi script over and over, allowing the agi to actually place the call to the party already on the phone, once the party called from the agi hangs up the process will repeat until phone a hangs up! I have writen some stuff in php that generates .call files and it so far seems to be solid. -Jonathan Jack Turer wrote: I am working on a web phone interface to give normal phonesets more 'virtual buttons'..etc, like the expensive executive phones via control via the web. This lead me to the following issue: I am wondering if it is possible (it doesn't seem to as far as I can tell) to make a script (AGI or otherwise) that will have asterisk automatically do the following without the user needing to originate any calls on their telephone: -Call an extension, hold on to the call (call A) -Call another extension, hold on to the call (call B) -Bridge the two calls (A and B) together (so the two extensions can talk to each other) -Later the script drops call B, but keeps call A up -Then asterisk calls another extension (call C) -Then asterisk bridges A with C so then they can talk to each other ..then later the same thing again (call D, then bridge with A)..etc...etc, Allthis would be AGI or script driven without any user having to press anything on his phone. Is this possible (the main issue I see in asterisk is that I cannot find a command in the asterisk API to bridge/unbridge calls like this without something being originated by a call into asterisk from a user). I looked at meetme, but it doesn't seem appropriate for what I want to do above. Any ideas? Thank you! Jack ___ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Advice on OS Choice
On Sat, 16 Oct 2004 20:34:21 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote: > I'm not sure which three letters you are talking about. My first > thought was "arse", but that's clearly four letters so you can't > mean that. "Mouth" is five letters long, so you can't mean that > either. Perhaps you mean "ear". Yes, that's probably it. :-) > > http://www.opensource.apple.com/darwinsource > > > There's something I didn't know. fair enough. > Thanks for pointing that out. > I'll take Apple off the "just like Apple and Microsoft" list then. It would seem Apple could improve their PR to make sure more people know about Darwin and the fact it's open source. Also, did you know that you can run Darwin natively on your x86 PC? I reckon this would be an easy way to verify and ensure that the current CVS compiles out of the box on MacOSX, but everybody I have told about this seemed rather surprised. > > The companies that should be singled out for bashing are those who > > violate the license terms, like Microsoft, who are using the BSD > > TCP/IP stack in Windoze without attribution to Berkely University; and > > SCO, who have still been distributing their own Linux distro while at > > the same time claiming that the GPL was null and void. > > > Yes. I thought that Apple were in the same camp, so I apologise for > that. The underlying points in my previous articles still stand - just > with Apple removed from the list. Fair enough. I don't necessarily have a problem with your views but I think it would help a great deal if you were to articulate them in a way that acknowledges that there are two different angles: one is legal the other is ethical >From what I gather reading the thread, a lot of the "heat" in the discussion would seem to stem from the boundaries between the legal and the ethical angle being somewhat unclear, if not mixed up. For example, let's assume for argument's sake that Apple had indeed taken the BSD code from open source to closed source. >From a legal angle, this would be OK as long as they also made the required attribution to Berkely University. One could not say that they had stolen the code. >From an ethical angle though the situation is much different. Ten years ago, it might have been ethically perfectly acceptable for any company to take the entire BSD code as a base for their own proprietary and closed source operating system as long as they abided by the BSD license, but in this day and age, our ethical standards have changed and any company using the BSD code would be expected to not only abide by the license, but also contribute something back. In this respect, Apple did the right thing, not only legally, but also ethically. Other companies, who have used the BSD code for their proprietary systems at a time when ethical standards were different, probably have done what was legally and ethically right at the time, but it might not feel right today. Fortunately, most of those companies have given back to community in other forms. For example, IBM have contributed a lot of code to Linux. Thus, in most cases there will still be a balance between how much a company has received and how much they have given back, thus they are still working within the boundaries of ethical standards even as those ethical standards are changing. So when you say that somebody is "stealing the code" in relation to the BSD license, I believe what you really mean to say is that they are unethically taking advantage of a gift without giving enough back in return. Not everybody will be able to read between the lines though and if you say "stealing" then it will not only lead to confusion but it can also be offending. Thus, my advice would be to make a clear disctinction between what is the right thing to do legally and what is the right thing to do ethically. I think if you do that, then you will find that most people will agree with you on what the right thing to do ethically is and should be. As far as the BSD license is concerned, I don't think that it is any more inviting to anybody to take advantage and do the unethical thing as is the GPL. The evidence would seem to suggest something different. Companies who have used BSD code for proprietary systems have mostly done so at a time when the ethical standards were different and as those standards changed, they have contributed back in other ways. Companies like Apple who have come to the BSD party much later, have done the ethically right thing and open sourced their improvements even though the license didn't mandate that. On the other hand, companies like Microsoft have no respect of the law nor ethics regardless what the license says. We only know that they are using BSD's TCP/IP stack from behavioural analysis. They didn't make the required attribution to Berkely University and never admitted that they are actually using the BSD code, so we have no way to tell other than through behavioural analysis. So if they have stolen t
[Asterisk-Users] X100P, Dutch analong line, caller-id
Dear all,
I've seen that there's progress to receive the caller-id over a KPN analog
phone line, but I can't get it to work on my X100P card. I've seen
Oliver's config
(http://www.mail-archive.com/[EMAIL PROTECTED]/msg53699.html)
but that doesn't yield any id.
Here my questions:
- Has anybody managed with a X100P?
- Is it enough for my inital debugging if I put in extensions.conf a line
similar to 'exten => s,1,NoOp,${CALLERID}' ?
Thanks in advance for any help or comments,
Max
P.S. That's the zapata.conf file I'm using to test:
[channels]
echocancel=yes
echocancelwhenbridged=yes
txgain=5%
context=incoming
signalling=fxs_ks
immediate=yes
usecallerid=yes
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no
callerid=asreceived
relaxdtmf=yes
useincomingcalleridonzaptransfer=yes
channel=>1
context=internal
signalling=fxo_ks
channel=>2
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Re: [Asterisk-Users] Asterisk Data Configuration Example 1
- Original Message - From: "Brian" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, October 16, 2004 6:53 PM Subject: Re: [Asterisk-Users] Asterisk Data Configuration Example 1 | You are more then welecome to edit the page and fix it. It is a WIKI | after all. Simply sign up for an account, then click the little "EDIT" | button at the top of the page you wish to edit. | | -Brian | I would, but 1. I have yet to get the T100P card to work at all, so anything I would add there is next to useless, I just know from working with the commands, either there are a bunch of unnecessary iterations or it is not clear what was trying to be accomplished. 2. I don't think my knowledge of the product is extensive enough to make a comment there. Regards Greg ___ 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] IAX UDP packet dropped on incoming call
It's probably because the hole in your firewall has closed. Either increase the amount of outbound traffic through that port (thus keeping the association alive), or modify your firewall to have a fixed port mapping to your asterisk box. -brian Gene Willingham wrote: When receiving incoming calls, I periodically get a UDP packet dropped message on my firewall. This prevents the incoming call from completing. It appears to be a random occurrence, sometimes hours, sometimes half hour, sometimes minutes. I am using Asterisk 1.0.1 if this helps. Gene ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P, Dutch analong line, caller-id
Hi,
> -Original Message-
> I've seen that there's progress to receive the caller-id over
> a KPN analog phone line, but I can't get it to work on my
> X100P card. I've seen Oliver's config
> (http://www.mail-archive.com/[EMAIL PROTECTED]/m
> sg53699.html)
> but that doesn't yield any id.
>
> Here my questions:
> - Has anybody managed with a X100P?
>
> - Is it enough for my inital debugging if I put in
> extensions.conf a line similar to 'exten => s,1,NoOp,${CALLERID}' ?
I havent' tried this with X100P cards recently, but I was under the
impression this would only be supported in the default codebase with TDM
cards.
Anyway, Dutch lines get callerid between first and second ring, so adding a
brief 'Wait' in your dialplan might help a bit.
Best regards,
Florian
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Re: [Asterisk-Users] Bandwidth control on a home office network
Hello: Try "ipfw and dummynet" on a freebsd box acting as traffic shapper, you can put your ATAs or like in another network, then you can manage your bandwith like as your ISP. If you need some assistance don't hessitate ask me Excuse me by the offtopic --- Ing. Julio Alvarez Tejera Unix Trends *BSD, Solaris & Linux CT & VoIP Solutions Finder (506) 286-5478 +1-305-704-2019 --- "estabilidad al extremo" - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, October 16, 2004 12:25 PM Subject: Re: [Asterisk-Users] Bandwidth control on a home office network > > I have a Grandstream ATA today connected to my 750k broadband > > connection via an older router / firewall that doesn't have any QoS / > > ToS capability. It works fine apart from the obvious problem of when > > large emails come in or somebody else on the network starts d/l-ing > > something big off the web. > > > > I'm wondering whether to swap the router for a Cisco in order to > > introduce some local bandwidth control. > > > > Alternatively I was wondering if I picked up a Cisco 7960 handset > > instead - is the 2nd ethernet port routed through the device, or does > > it just act as an Ethernet repeater, i.e. if I arranged the handset in > > the network as below would I get bandwidth prioritisation for the 7960? > > > > [CABLE MODEM]--[7960]---[FW / ROUTER / HUB][REST OF MY > > NETWORK] > > QoS, regardless of whether its based on the IP header TOS bits or on > specific tcp/udp port numbers, essentially prioritizes the "outbound" > flow of data packets, sending high priority packets before lower > priority packets. It does nothing for inbound data such as downloads > to your site. > > Most broadband connections have a different upload vs download speed, > where usually the download speed is substantially greater then the > upload speed. E.g., not uncommon to see DSL or Cable modems limited > to 758k down and 128k/256k upload speeds. QoS may help with prioritizing > traffic through the 128k/256k. However, your internet service provider > would need to prioritize the download traffic for you. > > There are some rather expensive devices that you can install that will > rate limit both upload and download traffic. Those devices artifically > control the download traffic by withholding TCP acknowledgment packets, > etc. Not sure how effective they are though. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P, Dutch analong line, caller-id
Hi, I've tried on Florians suggestions to put in a 'Wait' statement, but unfortunately that didn't help, either. So is my only change to sell the X100P card and get myself a TDM400? What's the big difference in caller id detection there? Regards and thanks, Max -- Maximilien Collon ___ 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] * Server behind a firewall - How To
Thanks to everyone for their feedback. I appreciate it. I will give it a try on Monday when I get back to my lab. If you have it, please send more info Thanks very much -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Sunday, October 17, 2004 3:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * Server behind a firewall - How To > My * server is NAT'd behind a firewall. > What ports do I need to open to allow a Grandstream IP to connect to it > remotely? You should read the wiki pages given above, but here is what I've done on my linksys: 4569 --> * 5060 --> * 1-10100 --> * in rtp.conf rtpstart=1 rtpend=10100 in sip.conf externip=123.123.123.123 I think that's all I had to do. -- When a simple answer can be given, it makes searching the list easier. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IaXY MWI
Sorry, hate to be dense here. I already have a working system with SIP phones with working MWI. I added the IaXY and it works but with no MWI. I added the mailbox=XXX statement to the iaxy.conf context but no MWI on the phone. Does anyone have MWI working with the Iaxy? -Original Message- From: Wilson Pickett [mailto:[EMAIL PROTECTED] Sent: Sunday, October 17, 2004 3:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IaXY MWI On Sat, 16 Oct 2004 06:54:16 -0400, Kevin <[EMAIL PROTECTED]> wrote: > I tried the mailbox statement and it didn't work. Include the voicemail.conf context as in voicemail.conf [users] 1234 => ,Joe Blow,[EMAIL PROTECTED],, ; etc [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Zapata on PowerMac G4
Hi Nick, > > I thought I would convert my old PowerMac G4 into a Linux box / > Asterisk server, but I haven't been having a lot of success. > > Call signaling seems to work fine - detects phones going on/off hook > and phones ring when you dial them (from the console) but audio doesn't > seem to be working at all - no dial tone, doesn't detect DTMF. > > The console (using OSS/dma_powermac) also doesn't work very well. I > typed 'dial 600' into the console and the sound is very jumpy - 1 > second sound, followed by 4 seconds silence. Processor is running at > 90% idle. I have tested full-duplex OSS using audio tools and it seems > to work perfectly. > I use a vintage PM9600 with Asterisk 1.0, four X100p, under YellowDog 3.0.1. I do not use the console at all so I can comment, but the rest of asterisk works 100% well, much better than on a cheap ASROCK motherboard with 2.0ghz AthlonXP processor. -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] Alternatives to the T100Ps?
Michael, I usually read and don't do much posting, but I had to on this. I am really suprised to see your commnets, and wondered what is the basis ? We have had a dual Xenon with a quad port T1 card in production for 16 months processing as many as 20,000 messaging calls a day. The box has never crashed, the board has never crashed, we haven't even restarted asterisk much less upgraded the code. I have never take a Bit Error on my DMS-500 from a Digium card. This is only one of several "production boxes" but the story is the same on all of them. How in the heck does this equate to: "hardware, drivers, or both is pretty sketchy" ? I would suggest just the opposite. Mark and the boys have done a great job on all fronts. How many Cisco AS-5300's have that record ? I have 9 of them brand new and not a single one is my answer. If you haven't looked at Digium lately, look again. -Original Message- From: Michael Loftis [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 12, 2004 2:17 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Alternatives to the T100Ps? Since I'm pretty sure that either the hardware, drivers, or both is pretty sketchy I'm interested in hearing alternatives, non-digium, and non-zaptel kernel modules, for ISDN PRI with *. -- GPG/PGP --> 0xE736BD7E 5144 6A2D 977A 6651 DFBE 1462 E351 88B9 E736 BD7E ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HylaFAX v. spandsp
Thanks for all your efforts, Steve. Because I'm unable to configure multiple PCI modem cards on the fax server, I'll have to give spandsp a go. Never a dull moment around here. Mike Steve Underwood wrote: Michael Welter wrote: Can someone explain where we are with spandsp? Is it ready for a production environment? How much will one fax using spandsp load the processor on an * system? Thanks, Mike Some people have been using spandsp-0.0.1k in production systems for months. The main problems have been: Canon fax machines often gave problems HP machines are buggy, and with spandsp the bugs cause black streaks oon received documents. People get pages chopped in half. You won't really notice the load caused by one channel of spandsp's fax machine on most modern CPUs. I had 100 channels running on a dual Xeon at one point, but I haven't tested recent versions at high loads. With the latest spandsp-0.0.2pre4 The bug affecting Canons seems to be resolved. The software used to rely on the internals of libtiff to compress and decompress images. Now it uses its own code which works around buggy machines likes the HPs, and allows better reporting of the received data quality. Of all the pages getting chopped problems I received, one turned out to be a stability issue in the modems. This seems to be resoolved. Every other report of pages getting chopped, that I have investigated, has been due to timing problems in * or the hardware. This is probably about 30 examples now. There is nothing I can do about this. If spandsp does not work in these cases, neither will hardware modems and HylaFAX with the calls passing through *. Modems just cannot tolerate hiccups in the audio stream. The long delay between spandsp-0.0.1k and the recent spandsp-0.0.2prexxx versions has been due to the fact that spandsp-0.0.1k has been working pretty well for a lot of people. There wasn't a need for many minor releases. 0.0.2 is a fairly major update, with many areas of the software changing in major ways. Some of these are complete, and some are still works in progress. The modems have been reworked to get closer to the theoretically achievable performance on noisy lines. Their compute requirements are also a bit lower now. Fax compression and decompression has been added. An incomplete skeleton for class 1 modem operation, to work with HylaFAX, has been added. A nearly working V.17 modem has been added. This still needs improvements to the receiver carrier locking to make it work properly. A half finished V.22bis modem has been added. A couple of people asked for this, so allow integration of things like credit card validation machines with *. Most of these still sues V.22bis. Variable speed playback of speech files. This is something a lot of people ask for, but which is fairly useless in practice :-). Its just to play with, though. A number of little modules, such as V.8 processing, have been implemented, tested, and added to spandsp, but are not really being used right now. Regards, Steve ___ 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 -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Advice on OS Choice
Joe Greco [EMAIL PROTECTED] wrote: > Kevin Walsh wrote: > > Joe Greco [EMAIL PROTECTED] asked his mother to write: > > > Kevin Walsh wrote: > > > > *plonk* > > > > > > Plonk yourself, retard. > > > > > Haha. Nice comeback - that must have taken you ages to think up. > > > Now: Are you even aware of the meaning of "*plonk*"? > > If so, why did you see my reply? > > For the non-USENET folks here, "*plonk*" is supposed to be an indication > that the poster has been relegated to a kill file or other filter > mechanism... replying to someone after you've "plonked" them is a great > way to scream "I'm a newbie and I have no idea what plonk means, but it > SOUNDS cool, look at me, I'm so cool!" > I see that you've looked up the meaning of *plonk* now, and have probably realised that "plonk yourself" is not technically correct in that context. Replying to someone after they claimed to have *plonked* you is just a cowardly way of ensuring that you get the last word in. I knew that you would either have no idea what the word meant, or would reply anyway, so I didn't bother with the filter at that time. Perhaps I should do so now. Now that we've sorted that out, can you please shut up. Your contrived medical examples have taken this thread so far away from Asterisk, it's unbelievable. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asked to transmit frame type 64, while native formats is 8
http://bugs.digium.com/bug_view_page.php?bug_id=0002519 If anyone has seen that error please come forward and report on this bug please. The original reporter is unwilling or unmotivated to even make an effort to assist in correcting the issue. So if anyone else has seen this please post. Thanks, Brian ___ 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] Simple phone question
Thanks for all the info Joe. I would really like to avoid cisco phones if possible. We've been bitten so many times by them already (Cisco Routers, Access Routers, Switches). Generally they provide no support without a contract, and even that does not keep them from end-of-lifing their products every year (or it least that's what happened with the brand new gear we've bought from them so far.. i.e. as5200, as5350, catalyst3500xl). I'd prefer someone else who will provide firmware fixes/updates without a contract. Say, where's the wiki? Jonathan Miller > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Joe Greco > Sent: Sunday, October 17, 2004 1:45 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Simple phone question > > > The real problem is that the old comdial is dying > > That's bad. Always a good motivation. We've been having BRI > TA's die, pushing us to VoIP too. ;-) > > > and cannot support any more extensions. > > That's bad too. > > > Not to mention constant static problems on speakerphone. > > The Cisco speakerphone is sweet. If you're looking for that > feature, be careful with other vendors. I hear there are > some good ones and some horribly bad ones. There's a lot of > helpful info on the WIKI and on this list. > > > (even after repunching the all the cables + replacing > them). We just > > really want a new system that will be expandable for the future. > > Assuming you can get used to a slightly different paradigm > than what you are used to, Asterisk can deliver that without a doubt. > > ... JG > -- > Joe Greco - sol.net Network Services - Milwaukee, WI - > http://www.sol.net "We call it the 'one bite at the apple' > rule. Give me one chance [and] then I won't contact you > again." - Direct Marketing Ass'n position on e-mail spam(CNN) > With 24 million small businesses in the US alone, that's way > too many apples. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_skinny usage of callerid
after finding out that chan_sccp is super buggy on my 12sp (ie hitting the wrong key crashes asterisk phone reboots , doesn't ring etc.) I went back to using chan_skinny this seems to be stable but the only thing that is not working that i need is caller id to be displayed on the phone when it rings. Anyone know if this works with chan_skinny ? any help would be great. Jason ___ 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] Bandwidth control on a home office network
On Sat, 16 Oct 2004 21:00:31 +0200, gramels wrote: >you might consider http://m0n0.ch/wall on a soekris.com or >pcengines.ch board which does nice trafficshaping for little money. >m0n0wall is a freebsd based opensource firewall appliance I heartily concur! I used m0n0wall on a Soekris 4501 to replace a Linksys BEFSR-81. m0n0 is a joy to use. You can try the PC version that ony requires a dual NIC'd old PC as a testbed to get you started. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 "Stay calm. Be brave. Watch for the signs." - Anne Sloboda, on the occasion of my wedding ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] Asked to transmit frame type 64, whilenative formats is 8
Care to post your findings to the bug note? Thanks, Brian > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Danny Froberg > Sent: Sunday, October 17, 2004 11:53 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asked to transmit frame type 64, whilenative > formats is 8 > > Think i solved that one by the ordering of allow= in sip.conf > > At 18:39 2004-10-17, you wrote: > >http://bugs.digium.com/bug_view_page.php?bug_id=0002519 > > > >If anyone has seen that error please come forward and report on this bug > >please. The original reporter is unwilling or unmotivated to even make > an > >effort to assist in correcting the issue. So if anyone else has seen > this > >please post. > > > >Thanks, > >Brian > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN
I am still in a planning phase. I do have a Panasonic PABX with an ISDN card. Will I be able to connect it to an asterisk machine using a normal ISDN card ? Felix ___ 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] IAXy setup
On Sun, 2004-10-17 at 00:20 -0400, Jim Van Meggelen wrote: > > Nevertheless, > it's kinda not proper to deliver an ethernet device that is not labeled > with it's MAC address. Why should we have to go through any kind of > trouble to determine this? I say it should be on the unit. The MAC address label should have the MAC address encoded as a bar code as well. When you're deploying hundreds of devices that and need to program their MAC addresses into your * box (or whatever) it helps to have the bar codes and a bar code reader. Jeff Ollie signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip to pstn gateway
Hello, here is what I am able to do: - I am able to register a SIP Phone on an Asterisk server. - I am able to call an extension on the remote Asterisk server with my SIP phone and hear the congratulation message. Informations about the configuration: - There is no phonecard (no digium card, no isdn card ...) in the Asterisk box but only a network interface card which connects the Asterisk server to a softswitch. This softswitch is the gateway to PSTN. Unfortunately I do not know anything about the softswitch. Is there something important that I should know about it? Here is what I would like to have: - I would like to be able to call a PSTN/ISDN phone with my SIP phone. - That means when I take my SIP Phone and dial a telephone number that belongs to the PSTN Asterisk must route the SIP packages to the softswitch which in turn routes the call to the PSTN. - When I dial another registered SIP phone Asterisk should connect the two sip phones so that they can speak to each other. - -- | SIP phone | ---> | Asterisk | ---> | Softswitch | ---> | PSTN | - -- | | - |> | Sip phone | - I have no idea how to configure Asterisk to accomplish this task. I started reading documents like ftp://ftp.isi.edu/in-notes/rfc3372.txt and the sip RFC and Mailing List articles and so on but they did not make me able to configure Asterisk in that way. Does anybody know where I can find documents that describe how I can do what I would like to have? Do I have to configure a SIP Proxy (SER for example) on the Asterisk box or does it work without a SIP proxy as well? Do I have to register the Asterisk box on the softswitch? (Should this be possible at all?) Thanks very much in advance for any kind of help. Regards, Fabian Müller ___ 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] IAXy setup
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Wilson Pickett > Sent: October 17, 2004 2:58 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] IAXy setup > > > Butting in because I own one too: You're *not* butting in! This is a community. Everyone's comments are welcome and valuable. > > 1. The MAC address needs to be visible on the unit. > Yes, Mark & Co, this is a good idea. > > > 2. DNS support. The IAXy needs to be able to handle names. > Too much to ask in such a simple device. Even though we'd all > like to see it. It may be a simple device, but it's also a $100 device. I can buy many devices for far less than $100 that can handle DNS (just look at your average SOHO router/switch). Keep in mind that if it was a SIP ATA it would have to support DNS, and those can be had for roughly the same price as an IAXy. I don't think DNS is too much to ask for at all. > > 3. Restore to factory. > Yes, please. I almost paralyzed my "paper clip hand" before I learned > that the reset button was only there for "aesthetic purposes". Yeah, that kinda blew my mind. > > 4. Some kind of TFTP, SSH or whatever is needed to allow > connection and > > configuration of the device. > > I can see why this is not the case. However, if some kind soul would > make a Windows command line "iaxyprovision.exe" I'd be happy. Sorry, but I CAN'T see why this is the case. Again, I'm back to my argument that there's plenty of devices in the same price range that have all kinds of administrative flexibility. Think about this: why is the IAXy stuck using a non-standard administrative interface? That is very uncommon in the world of networking hardware. I would like to think that the IAXy is built on a pretty flexible platform. For example, the FXS card in it is obviously the same one available for the TDM400. If that is so, it seems that (physically at least) the IAXy would also be capable of handling the FXO daughtercard. I'm hoping we'll be seeing an FXO IAXy at some point. I am aware that what I am asking is not necessarily simple. I am not suggesting that it'd be easy to implement the changes I'm suggesting; but easy or not, the IAXy needs to improve its functionality. It's a good first effort . . . no, make that a great first effort. As has been noted before: what it does, it does extremely well, with no real bugs to speak of. That is an inspiring accomplishment, and is no small part of the reason why I am very enthusiastic about its future. But I feel that it is still much more a prototype than something that is optomized for wide-scale deployment. It's very much a matter of opinion, I suppose. Cheers, -- Jim > ___ > 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 > > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 > > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX error messages
I just setup * to work with FWD and I'm now seeing these error messages: IAX Packet 31216 from circuit ids 212->1conflicts with earlier call with circuit ids 1->124 What can be causing this? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ 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] IAXy setup
Sorry, but I CAN'T see why this is the case. Again, I'm back to my argument that there's plenty of devices in the same price range that have all kinds of administrative flexibility. Think about this: why is the IAXy stuck using a non-standard administrative interface? That is very uncommon in the world of networking hardware. The IAXy has 4k (or is it 2k?) of RAM and 4k (or maybe 2k) of FLASH. Even in assembly its tough to do VoIP in 4k and in my opinion impossible to do VoIP, DNS, and HTTP server in 4k. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Advice on OS Choice
> Joe Greco [EMAIL PROTECTED] wrote: > > Kevin Walsh wrote: > > > Joe Greco [EMAIL PROTECTED] asked his mother to write: > > > > Kevin Walsh wrote: > > > > > *plonk* > > > > > > > > Plonk yourself, retard. > > > > > > > Haha. Nice comeback - that must have taken you ages to think up. > > > > > Now: Are you even aware of the meaning of "*plonk*"? > > > > If so, why did you see my reply? > > > > For the non-USENET folks here, "*plonk*" is supposed to be an indication > > that the poster has been relegated to a kill file or other filter > > mechanism... replying to someone after you've "plonked" them is a great > > way to scream "I'm a newbie and I have no idea what plonk means, but it > > SOUNDS cool, look at me, I'm so cool!" > > I see that you've looked up the meaning of *plonk* now, and have > probably realised that "plonk yourself" is not technically correct in > that context. Yes, it is. It suggests that you go killfile yourself, sparing us the tediousness. > Replying to someone after they claimed to have *plonked* you is just > a cowardly way of ensuring that you get the last word in. *plonk*ing someone is a cowardly way of trying to say "I am trying to win by virtue of trying to make my message the last word." I fail to be impressed; PKB rule clearly applies, as do some others. > I knew that > you would either have no idea what the word meant, or would reply anyway, > so I didn't bother with the filter at that time. Perhaps I should do so > now. Uh huh. Actually, it was I who recognized you as an old time alt.flame luser, and I just couldn't resist seeing if you were one of those sissies who likes to say "*plonk*" but who can't follow through on the requirement to actually killfile the person and stop replying to further messages - a fundamental requirement when you "*plonk*". > Now that we've sorted that out, can you please shut up. Your contrived > medical examples have taken this thread so far away from Asterisk, it's > unbelievable. My "contrived medical example" is currently a product manufactured by a global household name that's installed in a nearby hospital to which you may someday find yourself hooked up to. HTH, HAND. Spare yourself the embarrassment of any further public errors. Best, ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Advice on OS Choice
GENTLEMEN: Can you not see what you're doing to your standing on this list, given that your discussion has devolved from any shred of technical content into a pure pissing contest? Please desist. B. ___ 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] Simple phone question
> Thanks for all the info Joe. I would really like to avoid cisco > phones if possible. We've been bitten so many times by them already (Cisco > Routers, Access Routers, Switches). Generally they provide no support > without a contract, and even that does not keep them from end-of-lifing > their products every year (or it least that's what happened with the brand > new gear we've bought from them so far.. i.e. as5200, as5350, > catalyst3500xl). Yeah, that's a problem, for sure. It's a "no one ever got fired for buying IBM" kind of thing. Actually, I'm fairly impressed with the 7960 though. > I'd prefer someone else who will provide firmware > fixes/updates without a contract. That'd be, let's see, um, oh, "everyone except Cisco" I think. My own reading suggests that Polycom would be the next thing to try. They have a number of phones roughly equivalent to the various Ciscos, followed fairly closely by the Snom phones. You will, of course, hear other opinions. > Say, where's the wiki? You haven't found the WIKI? http://www.voip-info.org ... *Very* useful. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net "We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] IAXy setup
Well, that certainly poses a problem. You're kidding, right? Ouch. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Eric Wieling > Sent: October 17, 2004 2:20 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] IAXy setup > > > > > Sorry, but I CAN'T see why this is the case. Again, I'm back to my > > argument that there's plenty of devices in the same price > range that > > have all kinds of administrative flexibility. Think about > this: why is > > the IAXy stuck using a non-standard administrative > interface? That is > > very uncommon in the world of networking hardware. > > The IAXy has 4k (or is it 2k?) of RAM and 4k (or maybe 2k) of FLASH. > Even in assembly its tough to do VoIP in 4k and in my opinion > impossible > to do VoIP, DNS, and HTTP server in 4k. > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 > > > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ 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] IAXy setup
> > > > > Sorry, but I CAN'T see why this is the case. Again, I'm back to my > > > argument that there's plenty of devices in the same price > > range that > > > have all kinds of administrative flexibility. Think about > > this: why is > > > the IAXy stuck using a non-standard administrative > > interface? That is > > > very uncommon in the world of networking hardware. I guess you dont use telnet on many Cisco routers do you? ___ 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] ISDN
On Sun, 17 Oct 2004, Lancia Ersatzteilservice C.C. wrote: > I am still in a planning phase. > I do have a Panasonic PABX with an ISDN card. > Will I be able to connect it to an asterisk machine using a normal ISDN card Probably. It is hard to find someone who has tried your particular combination if you don't give us more details. Which exact model? Which isdn card in the pabx? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HandyTone 486 vs. Iaxy
Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com I know everyone on the list would prefer we all buy Digium and I also believe we should support them whenever we can. Anyhow, I was hoping for a somewhat unbiased opinion on the pros and cons on these two devices. We are looking for cheap but most importantly I want to know if anyone has seen one perform better than the other voice wise or are they all about the same when it comes to voice quality? I am quite the newbie here as you might imagine from my comments. I am leaning towards the IAXy because it supports IAX and I figured that was a good thing. However, if we are going to buy a lot of these at some point I think the price difference vs. features on the HandyTone 486 is at least something we should consider. I like the fact that the HandyTone has 2 ports, does this mean I can configure two different phone numbers, one for each port or is it for tow outgoing lines only? Also, I can offer someone a second phone line without any additional equipment etc. Is the fact that the IAXy supports IAX a BIG factor? Will Sip work just as well with Asterisk? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com ___ 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] HandyTone 486 vs. Iaxy
SIP will give you hell with NAT, IAX Wont. Thank you, Steve Maroney On Sun, 17 Oct 2004, Your Own ISP .com wrote: > > > Thanks, > Todd Routhier > Lightwave Technologies, LLC. > > -- > Start Your Dialup Internet Service! > http://www.YourOwnISP.com > > > Lightwave Technologies, LLC. > http://www.LightWaveTech.com > > I know everyone on the list would prefer we all buy Digium and I also > believe we should support them whenever we can. > > Anyhow, I was hoping for a somewhat unbiased opinion on the pros and cons on > these two devices. > > We are looking for cheap but most importantly I want to know if anyone has > seen one perform better than the other voice wise or are they all about the > same when it comes to voice quality? > > I am quite the newbie here as you might imagine from my comments. > > I am leaning towards the IAXy because it supports IAX and I figured that was > a good thing. However, if we are going to buy a lot of these at some point I > think the price difference vs. features on the HandyTone 486 is at least > something we should consider. > > I like the fact that the HandyTone has 2 ports, does this mean I can > configure two different phone numbers, one for each port or is it for tow > outgoing lines only? Also, I can offer someone a second phone line without > any additional equipment etc. > > Is the fact that the IAXy supports IAX a BIG factor? Will Sip work just as > well with Asterisk? > > > > Thanks, > Todd Routhier > Lightwave Technologies, LLC. > > -- > Start Your Dialup Internet Service! > http://www.YourOwnISP.com > > > Lightwave Technologies, LLC. > http://www.LightWaveTech.com > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HandyTone 486 vs. Iaxy
Really, I have been using Cisco ATA 186's for over a year behind NATS, I thought these were Sip, maybe not. These have been working pretty good.. Thanks for the feedback though.. Oh, sorry about my signature appearing at the top of my last message, I don't even know how the *** that happen. Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Maroney Sent: Sunday, October 17, 2004 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HandyTone 486 vs. Iaxy SIP will give you hell with NAT, IAX Wont. Thank you, Steve Maroney On Sun, 17 Oct 2004, Your Own ISP .com wrote: > > > Thanks, > Todd Routhier > Lightwave Technologies, LLC. > > -- > Start Your Dialup Internet Service! > http://www.YourOwnISP.com > > > Lightwave Technologies, LLC. > http://www.LightWaveTech.com > > I know everyone on the list would prefer we all buy Digium and I also > believe we should support them whenever we can. > > Anyhow, I was hoping for a somewhat unbiased opinion on the pros and cons on > these two devices. > > We are looking for cheap but most importantly I want to know if anyone has > seen one perform better than the other voice wise or are they all about the > same when it comes to voice quality? > > I am quite the newbie here as you might imagine from my comments. > > I am leaning towards the IAXy because it supports IAX and I figured that was > a good thing. However, if we are going to buy a lot of these at some point I > think the price difference vs. features on the HandyTone 486 is at least > something we should consider. > > I like the fact that the HandyTone has 2 ports, does this mean I can > configure two different phone numbers, one for each port or is it for tow > outgoing lines only? Also, I can offer someone a second phone line without > any additional equipment etc. > > Is the fact that the IAXy supports IAX a BIG factor? Will Sip work just as > well with Asterisk? > > > > Thanks, > Todd Routhier > Lightwave Technologies, LLC. > > -- > Start Your Dialup Internet Service! > http://www.YourOwnISP.com > > > Lightwave Technologies, LLC. > http://www.LightWaveTech.com > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can not compile chan_capi 0.3.5
Hello, i can not compile chan_capi 0.3.5 on a suse 9.1 plattform. i run latest asterisk cvs build 14/10/04. just type make and become: # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from /usr/include/linux/kernelcapi.h:13, from /usr/include/linux/capi.h:18, from chan_capi.c:35: /usr/include/linux/list.h:604:2: warning: #warning "don't include kernel headers in userspace" chan_capi.c: In function `capi_new': chan_capi.c:1073: error: structure has no member named `callerid' chan_capi.c:1074: error: structure has no member named `dnid' chan_capi.c: In function `pipe_msg': chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c:1724: error: structure has no member named `dnid' chan_capi.c: In function `load_module': chan_capi.c:2793: warning: passing arg 4 of `ast_channel_register' from incompatible pointer type make: *** [chan_capi.o] Error 1 # A google search can not help. can you help me ? greetings nicolas ___ 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] HandyTone 486 vs. Iaxy
Also, I see this in the description of the 486: Built-in router, NAT and Gateway Is it possible that this thing will prioritize the bandwidth in favor of the voice? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Your Own ISP .com Sent: Sunday, October 17, 2004 3:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] HandyTone 486 vs. Iaxy Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com I know everyone on the list would prefer we all buy Digium and I also believe we should support them whenever we can. Anyhow, I was hoping for a somewhat unbiased opinion on the pros and cons on these two devices. We are looking for cheap but most importantly I want to know if anyone has seen one perform better than the other voice wise or are they all about the same when it comes to voice quality? I am quite the newbie here as you might imagine from my comments. I am leaning towards the IAXy because it supports IAX and I figured that was a good thing. However, if we are going to buy a lot of these at some point I think the price difference vs. features on the HandyTone 486 is at least something we should consider. I like the fact that the HandyTone has 2 ports, does this mean I can configure two different phone numbers, one for each port or is it for tow outgoing lines only? Also, I can offer someone a second phone line without any additional equipment etc. Is the fact that the IAXy supports IAX a BIG factor? Will Sip work just as well with Asterisk? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk for a VOIP Provider?
Is it practical to use Asterisk as the basis for a VOIP provider service? What I am saying is that we want to start a VOIP service offering including possibly terminating/originating some of our own traffic at some point as we grow. How many simultaneous end users (single line) should we expect to service with say one Asterisk box with say 1 T1 for data, hardware would be 2.4 ghz or so with a gig of ram. We would be using sip or Iax adapters (which would be better bandwidth wise) if that helps. Just trying to get an idea of what sort of resources we need to get ourselves rolling. Lastly, this is allowed within the license of Asterisk right? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com ___ 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] HandyTone 486 vs. Iaxy
On Sun, 2004-10-17 at 15:03 -0500, Your Own ISP .com wrote: > > Thanks, > Todd Routhier > Lightwave Technologies, LLC. > > -- > Start Your Dialup Internet Service! > http://www.YourOwnISP.com > > > Lightwave Technologies, LLC. > http://www.LightWaveTech.com Did we need to be spammed? Do you know what lazy foul you commited when I tell you I use a threaded mail reader? > Is the fact that the IAXy supports IAX a BIG factor? Will Sip work just as > well with Asterisk? SIP isn't as good of a protocol as IAX, so hedge your bet on support versus per port cost. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk for a VOIP Provider?
Dude, go learn to use your email program. Turn off HTML. On Sun, 2004-10-17 at 15:14 -0500, [EMAIL PROTECTED] wrote: > Is it practical to use Asterisk as the basis for a VOIP provider > service? > > What I am saying is that we want to start a VOIP service offering > including possibly terminating/originating some of our own traffic at > some point as we grow. > > How many simultaneous end users (single line) should we expect to > service with say one Asterisk box with say 1 T1 for data, hardware > would be 2.4 ghz or so with a gig of ram. We would be using sip or Iax > adapters (which would be better bandwidth wise) if that helps. The fact that you asked the question about how many users you could support on a T1 means you really need to go study standard telephony service for a while. This is regardless of Voip or not. > Just trying to get an idea of what sort of resources we need to get > ourselves rolling. Several books and many hours to familiarize yourself with telephony, then you need to get a lawyer to tell you what your local and federal laws will require of you. > Lastly, this is allowed within the license of Asterisk right? Yep, but your local regulators will be your biggest hurdle. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can not compile chan_capi 0.3.5
On Sun, 2004-10-17 at 22:12, Nicolas wrote: > Hello, > > i can not compile chan_capi 0.3.5 on a suse 9.1 plattform. > i run latest asterisk cvs build 14/10/04. Try the stable branch: cvs checkout -r v1-0 zaptel libpri asterisk Works for me. Regards, Patrick ___ 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] Asterisk for a VOIP Provider?
As you can see in my previous post I normally do before sending to the list, I forgot this time, My Apologies. Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Sunday, October 17, 2004 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk for a VOIP Provider? Dude, go learn to use your email program. Turn off HTML. On Sun, 2004-10-17 at 15:14 -0500, [EMAIL PROTECTED] wrote: > Is it practical to use Asterisk as the basis for a VOIP provider > service? > > What I am saying is that we want to start a VOIP service offering > including possibly terminating/originating some of our own traffic at > some point as we grow. > > How many simultaneous end users (single line) should we expect to > service with say one Asterisk box with say 1 T1 for data, hardware > would be 2.4 ghz or so with a gig of ram. We would be using sip or Iax > adapters (which would be better bandwidth wise) if that helps. The fact that you asked the question about how many users you could support on a T1 means you really need to go study standard telephony service for a while. This is regardless of Voip or not. > Just trying to get an idea of what sort of resources we need to get > ourselves rolling. Several books and many hours to familiarize yourself with telephony, then you need to get a lawyer to tell you what your local and federal laws will require of you. > Lastly, this is allowed within the license of Asterisk right? Yep, but your local regulators will be your biggest hurdle. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] HandyTone 486 vs. Iaxy
Again, it was an oversight.. Again I apologize to you and the list.. I actually mentioned that I had not idea why my signature ended up first. My signature is configured the same for all emails I send, if it's offensive to the list I will try my best to trim it down or something before posting. Maybe you just didn't like that it was at the top, if so that was an oversite on my part. Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Sunday, October 17, 2004 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HandyTone 486 vs. Iaxy On Sun, 2004-10-17 at 15:03 -0500, Your Own ISP .com wrote: > > Thanks, > Todd Routhier > Lightwave Technologies, LLC. > > -- > Start Your Dialup Internet Service! > http://www.YourOwnISP.com > > > Lightwave Technologies, LLC. > http://www.LightWaveTech.com Did we need to be spammed? Do you know what lazy foul you commited when I tell you I use a threaded mail reader? > Is the fact that the IAXy supports IAX a BIG factor? Will Sip work > just as well with Asterisk? SIP isn't as good of a protocol as IAX, so hedge your bet on support versus per port cost. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] IAXy setup
Telnet is a standard, is it not? > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Steve Totaro > Sent: October 17, 2004 2:58 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] IAXy setup > > > > > > > > > > Sorry, but I CAN'T see why this is the case. Again, I'm > back to my > > > > argument that there's plenty of devices in the same price > > > range that > > > > have all kinds of administrative flexibility. Think about > > > this: why is > > > > the IAXy stuck using a non-standard administrative > > > interface? That is > > > > very uncommon in the world of networking hardware. > > > I guess you dont use telnet on many Cisco routers do you? > > > ___ > 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 > > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 > > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wildcard X100P and Fedora Core 2?
I have read in the Wiki about having to recompile the kernel and such to be able to use this card in Asterisk. I have not attempted any of the special steps mentioned yet but I do have the card installed in a slot. I installed Asterisk and have it up and running except for the FXO card mentioned above. I noticed that when I start * at the command line with the -vvv I see that Asterisk seems to recognize some sort of card that mentions a Best Data chip set (or something like that). Is it possible that the card will just work without the Kernel adjustments mentioned for Fedora 2? Has anyone been able to get it to work without the Kernel changes? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: can not compile chan_capi 0.3.5
Patrick wrote: > On Sun, 2004-10-17 at 22:12, Nicolas wrote: >> Hello, >> >> i can not compile chan_capi 0.3.5 on a suse 9.1 plattform. >> i run latest asterisk cvs build 14/10/04. > > Try the stable branch: cvs checkout -r v1-0 zaptel libpri asterisk > Works for me. > > Regards, > Patrick > > ___ > 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 Thanks but asterisk is working, my problem is the chan_capi channel driver. nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: can not compile chan_capi 0.3.5
Hello, Patrick is correct. In CVS head the callerid related stuff underwent a huge reworking. This rendered compatibility with previous source non-existant. Thus, chan_capi can not be used with CVS head. The callerid change did not occur in stable and thus chan_capi will work with stable. Joshua Colp Senior Software Developer VoiceConduits, LLC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Sent: Sunday, October 17, 2004 1:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: can not compile chan_capi 0.3.5 Patrick wrote: > On Sun, 2004-10-17 at 22:12, Nicolas wrote: >> Hello, >> >> i can not compile chan_capi 0.3.5 on a suse 9.1 plattform. >> i run latest asterisk cvs build 14/10/04. > > Try the stable branch: cvs checkout -r v1-0 zaptel libpri asterisk > Works for me. > > Regards, > Patrick > > ___ > 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 Thanks but asterisk is working, my problem is the chan_capi channel driver. nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX 0.9.9b - now multi codec support
Hi all, Thanks to the great work of Steve Kann on the iaxclient library, now DIAX is able to support the following codecs: - uLaw (still a little ptoblem with the sound in one direction) - GSM - iLBC - Speex You can download version 0.9.9b from the following address: http://www.geocities.com/tdanro/diax/diax099b.zip The help file and the web page is not yet updated (I work on this now). For the latest available help file use the address: http://www.laser.com/dante/diax/diaxhlp.htm Please play with it and send me your feedback. It is not fully tested, so...please be carefull. Thank you for your help and best regards, Dan P.S. The updated source file for the wiax.dll will be available soon on my site. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: can not compile chan_capi 0.3.5
On Sun, 2004-10-17 at 22:58, Nicolas wrote: > Patrick wrote: > > > On Sun, 2004-10-17 at 22:12, Nicolas wrote: > >> Hello, > >> > >> i can not compile chan_capi 0.3.5 on a suse 9.1 plattform. > >> i run latest asterisk cvs build 14/10/04. > > chan_capi uses header files from asterisk. Look in the chan_capi Makefile and you will see. Obviously chan_capi does not know about the new callerid code that is part of recent asterisk cvs. They are tied together. That is why you need to use v1-0 of asterisk or wait until kapejod releases an updated chan_capi (prepare for a wait afaik). Or fix it yourself off course... Regards, Patrick ___ 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] HandyTone 486 vs. Iaxy
Also wanted to throw the Sipura SPA-2000 into the ring and see what you all think of that. Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Your Own ISP .com Sent: Sunday, October 17, 2004 3:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] HandyTone 486 vs. Iaxy Also, I see this in the description of the 486: Built-in router, NAT and Gateway Is it possible that this thing will prioritize the bandwidth in favor of the voice? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Your Own ISP .com Sent: Sunday, October 17, 2004 3:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] HandyTone 486 vs. Iaxy Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com I know everyone on the list would prefer we all buy Digium and I also believe we should support them whenever we can. Anyhow, I was hoping for a somewhat unbiased opinion on the pros and cons on these two devices. We are looking for cheap but most importantly I want to know if anyone has seen one perform better than the other voice wise or are they all about the same when it comes to voice quality? I am quite the newbie here as you might imagine from my comments. I am leaning towards the IAXy because it supports IAX and I figured that was a good thing. However, if we are going to buy a lot of these at some point I think the price difference vs. features on the HandyTone 486 is at least something we should consider. I like the fact that the HandyTone has 2 ports, does this mean I can configure two different phone numbers, one for each port or is it for tow outgoing lines only? Also, I can offer someone a second phone line without any additional equipment etc. Is the fact that the IAXy supports IAX a BIG factor? Will Sip work just as well with Asterisk? Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone else tried Speex 1.1 CVS?
I built the CVS version of the Speex library - v1.2 it calls itself. Asterisk seg faults trying to use codec_speex.so. I'll have a look to try to fix it, but thought I'd just ask if anyone else knows what needs to be done? Steve ___ 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] Anyone else tried Speex 1.1 CVS?
On Sun, 2004-10-17 at 23:16, [EMAIL PROTECTED] wrote: > I built the CVS version of the Speex library - v1.2 it calls itself. > Asterisk seg faults trying to use codec_speex.so. > I'll have a look to try to fix it, but thought I'd just ask if anyone else > knows what needs to be done? > Use version 1.0.4. Works on my box (v1-0). Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - new SPA-3000 firmware out (v2.0.11a)
A little off topic, but no where near as bad as the gpl discussion, but the spa-3000 has new firmware, v2.0.11a, at http://www.sipura.com/support/index.htm Release notes are rather short, but initial tests indicate its working well so far. Seems to have improved the small amount of echo that I've occasionally heard on earlier versions for the few calls that I've made thus far. ___ 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] Sending broadcasts to all phones?
I am in the process of writing an app to do this with Cisco phones7940/60. The feature on most PBX's is Page Groups, This allows paging through the speaker phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Saturday, October 16, 2004 5:36 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sending broadcasts to all phones? The Polycom phones will do this. Use the meetme feature. It's well documented on the Wiki. John David J Carter wrote: > I have a Panasonic switch here and it a paging system on the switch. > > It will output the page message to all phones and also to an RCA (Phono) > socket on the side of the switch to a PA amplifier if required to drive a > 100Volt line system around a building. > > Dave > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Kevin Walsh > Sent: 16 October 2004 22:28 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Sending broadcasts to all phones? > > > Kristian Kielhofner [EMAIL PROTECTED] wrote: > >>Stan Brinkerhoff wrote: >> >>>A friend of mine has a real panasonic PBX setup at his house, and is >>>able to pick up the phone, dial an extension, and it broadcasts what he >>>says over every phone in his house without the phones having to be >>>picked up. What is this feature called? >>> >>>Would it be possible to set this up with Asterisk given the appropriate >>>phones? (Cisco?) >>> >> >>This can be done with Cisco phones and 6.x or 7.x firmware. It is on >>the wiki. >> > > Well, actually, it's not on the WIKI. The WIKI would help you set up > a Cisco phone to auto-answer, but that's not all he needs here. > The problem is that if you dial "phone1&phone2" then the first phone > to auto-answer will receive the "broadcasted" call. The other phones > in the list will not hear anything. Well, that'd be what I'd expect > to happen with Dial(), anyway. > > Stan seems to be asking for a system where the caller hears a ring tone > until all phones (auto)answer, and is then able to speak to them all at > once. It'd be kind of like an "enforced conference call", but with one > speaker and multiple listeners, and with all audio received from the > called phones thrown away rather than distributed. > > It could be done, but would need a new Dial()-based application to do > it, I think. Perhaps there's an existing facility that can be used to > to do this. If there is then I can't think of it. > > -- >_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ > _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h > _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] > _/ _/ _/_/_/_/ _/_/_/_/ _/_/ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: can not compile chan_capi 0.3.5
The other thing you can do is to check out cvs head before the callerid rework was done. I would suggest a date like September 19th or so. I'm using myself CVS-HEAD-08/13/04 with chan_capi without problems. Kind regards, Martin List-Petersen On Mon, 2004-10-18 at 02:05, Joshua Colp wrote: > Hello, > > Patrick is correct. In CVS head the callerid related stuff underwent a huge > reworking. This rendered compatibility with previous source non-existant. > Thus, chan_capi can not be used with CVS head. The callerid change did not > occur in stable and thus chan_capi will work with stable. > > Joshua Colp > Senior Software Developer > VoiceConduits, LLC. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas > Sent: Sunday, October 17, 2004 1:59 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Re: can not compile chan_capi 0.3.5 > > Patrick wrote: > > > On Sun, 2004-10-17 at 22:12, Nicolas wrote: > >> Hello, > >> > >> i can not compile chan_capi 0.3.5 on a suse 9.1 plattform. > >> i run latest asterisk cvs build 14/10/04. > > > > Try the stable branch: cvs checkout -r v1-0 zaptel libpri asterisk > > Works for me. > > > > Regards, > > Patrick > > > > ___ > > 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 > > Thanks but asterisk is working, my problem is the chan_capi channel driver. > > nicolas > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA-186 and Caller ID
I'm having an interesting issue with the caller id generation of the Cisco ata-186. When the information is displayed, the name is displayed properly yet the number is corrupted, I get several solid boxes followed by a one. The ATA is set to the default bellcore as it recommends for use in the United States. Any suggestions on what to look into? Michael ___ 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] HandyTone 486 vs. Iaxy
The 486 acts as a router and DHCP server so the problems with SIP and NAT are negated if the 486 is used for that purpose (at least on the remote end) since its wan interface has a public ip. Thanks, Steve Totaro www.totartechnologies.com - Original Message - From: "Your Own ISP .com" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Sunday, October 17, 2004 4:34 PM Subject: RE: [Asterisk-Users] HandyTone 486 vs. Iaxy > Again, it was an oversight.. Again I apologize to you and the list.. I > actually mentioned that I had not idea why my signature ended up first. > > My signature is configured the same for all emails I send, if it's offensive > to the list I will try my best to trim it down or something before posting. > Maybe you just didn't like that it was at the top, if so that was an > oversite on my part. > > > Thanks, > Todd Routhier > Lightwave Technologies, LLC. > > -- > Start Your Dialup Internet Service! > http://www.YourOwnISP.com > > > Lightwave Technologies, LLC. > http://www.LightWaveTech.com > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Steven > Critchfield > Sent: Sunday, October 17, 2004 3:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] HandyTone 486 vs. Iaxy > > On Sun, 2004-10-17 at 15:03 -0500, Your Own ISP .com wrote: > > > > Thanks, > > Todd Routhier > > Lightwave Technologies, LLC. > > > > -- > > Start Your Dialup Internet Service! > > http://www.YourOwnISP.com > > > > > > Lightwave Technologies, LLC. > > http://www.LightWaveTech.com > > Did we need to be spammed? > > Do you know what lazy foul you commited when I tell you I use a threaded > mail reader? > > > > Is the fact that the IAXy supports IAX a BIG factor? Will Sip work > > just as well with Asterisk? > > SIP isn't as good of a protocol as IAX, so hedge your bet on support versus > per port cost. > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy setup
yup it is, just not a web interface like what I thought the writer was implying. I thought setting it up was a breeze. No complaints here. - Original Message - From: "Jim Van Meggelen" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Sunday, October 17, 2004 4:35 PM Subject: RE: [Asterisk-Users] IAXy setup > Telnet is a standard, is it not? > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Steve Totaro > > Sent: October 17, 2004 2:58 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] IAXy setup > > > > > > > > > > > > > > > Sorry, but I CAN'T see why this is the case. Again, I'm > > back to my > > > > > argument that there's plenty of devices in the same price > > > > range that > > > > > have all kinds of administrative flexibility. Think about > > > > this: why is > > > > > the IAXy stuck using a non-standard administrative > > > > interface? That is > > > > > very uncommon in the world of networking hardware. > > > > > > I guess you dont use telnet on many Cisco routers do you? > > > > > > ___ > > 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 > > > > > > --- > > Incoming mail is certified Virus Free. > > Checked by AVG anti-virus system (http://www.grisoft.com). > > Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 > > > > > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 486 vs. Iaxy
Good point ! Thank you, Steve Maroney On Sun, 17 Oct 2004, Steve Totaro wrote: > The 486 acts as a router and DHCP server so the problems with SIP and NAT > are negated if the 486 is used for that purpose (at least on the remote end) > since its wan interface has a public ip. > > Thanks, > Steve Totaro > www.totartechnologies.com > > > - Original Message - > From: "Your Own ISP .com" <[EMAIL PROTECTED]> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[EMAIL PROTECTED]> > Sent: Sunday, October 17, 2004 4:34 PM > Subject: RE: [Asterisk-Users] HandyTone 486 vs. Iaxy > > > > Again, it was an oversight.. Again I apologize to you and the list.. I > > actually mentioned that I had not idea why my signature ended up first. > > > > My signature is configured the same for all emails I send, if it's > offensive > > to the list I will try my best to trim it down or something before > posting. > > Maybe you just didn't like that it was at the top, if so that was an > > oversite on my part. > > > > > > Thanks, > > Todd Routhier > > Lightwave Technologies, LLC. > > > > -- > > Start Your Dialup Internet Service! > > http://www.YourOwnISP.com > > > > > > Lightwave Technologies, LLC. > > http://www.LightWaveTech.com > > > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Steven > > Critchfield > > Sent: Sunday, October 17, 2004 3:15 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] HandyTone 486 vs. Iaxy > > > > On Sun, 2004-10-17 at 15:03 -0500, Your Own ISP .com wrote: > > > > > > Thanks, > > > Todd Routhier > > > Lightwave Technologies, LLC. > > > > > > -- > > > Start Your Dialup Internet Service! > > > http://www.YourOwnISP.com > > > > > > > > > Lightwave Technologies, LLC. > > > http://www.LightWaveTech.com > > > > Did we need to be spammed? > > > > Do you know what lazy foul you commited when I tell you I use a threaded > > mail reader? > > > > > > > Is the fact that the IAXy supports IAX a BIG factor? Will Sip work > > > just as well with Asterisk? > > > > SIP isn't as good of a protocol as IAX, so hedge your bet on support > versus > > per port cost. > > > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: can not compile chan_capi 0.3.5
Or better yet fix it yourself its like all of a few lines to make it work. bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Patrick > Sent: Sunday, October 17, 2004 4:10 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [Asterisk-Users] Re: can not compile chan_capi 0.3.5 > > On Sun, 2004-10-17 at 22:58, Nicolas wrote: > > Patrick wrote: > > > > > On Sun, 2004-10-17 at 22:12, Nicolas wrote: > > >> Hello, > > >> > > >> i can not compile chan_capi 0.3.5 on a suse 9.1 plattform. > > >> i run latest asterisk cvs build 14/10/04. > > > > > chan_capi uses header files from asterisk. Look in the chan_capi > Makefile and you will see. Obviously chan_capi does not know about the > new callerid code that is part of recent asterisk cvs. They are tied > together. That is why you need to use v1-0 of asterisk or wait until > kapejod releases an updated chan_capi (prepare for a wait afaik). Or fix > it yourself off course... > > Regards, > Patrick > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXy setup
As with so many things, perspective determines impressions. If I look at the IAXy from the perspective of a guy in his lab with his Asterisk, then this thing sets up about as painlessly as anything could. I like it a lot. But if one looks at it from the perspective of a solutions provider, possibly needing to deploy these by the dozens, hundreds or thousands, a desire to optimize the process surfaces. The management interface to the IAXy does not lend itself to mass deployments, and that is the source of my criticism. I, personally, found the IAXy a breeze to configure. But when I look at them as a project manager or implementer, there are some glaring shortcomings that I'd love to see fixed. And I definitely misspoke if I gave the impression that I wanted a web interface. That doesn't interest me nearly as much as being able to assign the MAC address to a DHCP server, and letting the IAXy grab it's personality though a bootstrap mechanism. And some sort of telnet/SSH interface, even a really ugly hex-based one, or possibly some funky DTMF interface via the FXS port would be HUGE. It's a wish list, ya know? > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Steve Totaro > Sent: October 17, 2004 6:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] IAXy setup > > > yup it is, just not a web interface like what I thought the > writer was implying. I thought setting it up was a breeze. > No complaints here. > > > - Original Message - > From: "Jim Van Meggelen" <[EMAIL PROTECTED]> > To: "'Asterisk Users Mailing List - Non-Commercial > Discussion'" <[EMAIL PROTECTED]> > Sent: Sunday, October 17, 2004 4:35 PM > Subject: RE: [Asterisk-Users] IAXy setup > > > > Telnet is a standard, is it not? > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On > Behalf Of Steve > > > Totaro > > > Sent: October 17, 2004 2:58 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [Asterisk-Users] IAXy setup > > > > > > > > > > > > > > > > > > > > Sorry, but I CAN'T see why this is the case. Again, I'm > > > back to my > > > > > > argument that there's plenty of devices in the same price > > > > > range that > > > > > > have all kinds of administrative flexibility. Think about > > > > > this: why is > > > > > > the IAXy stuck using a non-standard administrative > > > > > interface? That is > > > > > > very uncommon in the world of networking hardware. > > > > > > > > > I guess you dont use telnet on many Cisco routers do you? > > > > > > > > > ___ > > > 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 > > > > > > > > > --- > > > Incoming mail is certified Virus Free. > > > Checked by AVG anti-virus system (http://www.grisoft.com). > > > Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 > > > > > > > > > > --- > > Outgoing mail is certified Virus Free. > > Checked by AVG anti-virus system (http://www.grisoft.com). > > Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/> asterisk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ 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] * Server behind a firewall - How To
Thanks for your feedback. What WiKi pages? I am not seeing any "ginen above". 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Sunday, October 17, 2004 3:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * Server behind a firewall - How To > My * server is NAT'd behind a firewall. > What ports do I need to open to allow a Grandstream IP to connect to it > remotely? You should read the wiki pages given above, but here is what I've done on my linksys: 4569 --> * 5060 --> * 1-10100 --> * in rtp.conf rtpstart=1 rtpend=10100 in sip.conf externip=123.123.123.123 I think that's all I had to do. -- When a simple answer can be given, it makes searching the list easier. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA-186 and Caller ID
On Sun, 17 Oct 2004 19:52:50 -0400, Cory Andrews <[EMAIL PROTECTED]> wrote: > Michael Greb wrote: > >I'm having an interesting issue with the caller id generation of the > >Cisco ata-186. > > > >When the information is displayed, the name is displayed properly yet > >the number is corrupted, I get several solid boxes followed by a one. > >The ATA is set to the default bellcore as it recommends for use in the > >United States. Any suggestions on what to look into? > > Michael - Which version of the Cisco ATA do you have, is is the I1 or I2 > version it should say on the back of the unit. l1, running firmware v3.1.0 atasip (Build 040211A) Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP outbound dialing -- newbie alert.
Hey, all. I've got a TDM400 with one each of FXO and FXS (currently unused) modules. I've also got a BudgeTone 101. The SIP side works fine (I can call myself, get dumped to voicemail, etc.), but I haven't yet figured out how to configure the FXO side of things -- leastwise, not properly. I read (for example) the Digium Asterisk guide, as well as the pretty good writeup on Onlamp -- but neither has a really dumb-as-dirt(tm) super-simple configuration. In other words, I'm looking for sample zaptel.conf, zapata.conf, and extensions.conf files that are INCREDIBLY simple, allowing outbound (and maybe even inbound ;-) calls, so I can get something working, and build from there. Any suggestions? If it means I need to go over docs I've already read, that's fine, but I'm pretty confused right now... Thanks, - Ken D'Ambrosio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_skinny callerID usage
after finding out that chan_sccp is super buggy on my 12sp (ie hitting the wrong key crashes asterisk phone reboots , doesn't ring etc.) I went back to using chan_skinny this seems to be stable but the only thing that is not working that i need is caller id to be displayed on the phone when it rings. Anyone know if this works with chan_skinny ? any help would be great. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax Redirection
Guys,
Can someone confirm this? Running CVS-HEAD-10/17/04-11:33:03.
Just noticed today that faxes are being detected and * debug shows
redirections to the dialplan fax extension but it does not get redirected.
Eventually the channel just times out. See debug log lines below. If I
edit the timeout line to manually redirect the fax, it works. Auto redirect
is what is broken.
auto.
-- Redirecting Zap/2-1 to fax extension
-- Timeout on Zap/2-1
manual...
-- Redirecting Zap/2-1 to fax extension
-- Timeout on Zap/2-1
== CDR updated on Zap/2-1
-- Executing Goto("Zap/2-1", "fax|2203|1") in new stack
-- Goto (fax,2203,1)
-- Executing Macro("Zap/2-1", "faxreceive") in new stack
-- Executing SetVar("Zap/2-1",
"FAXFILE=/var/spool/asterisk-fax/1098060806.2.tif") in new stack
-- Executing RxFAX("Zap/2-1",
"/var/spool/asterisk-fax/1098060806.2.tif") in new stack
--
Gary White [EMAIL PROTECTED]
Network Administrator Internet Pathway
105 D East Church Street Voice: 601-776-3355
P. O. Box 777 Fax: 601-776-2314
Quitman, MS 39355Registered Linux User Number 198875
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[Asterisk-Users] chan_skinny caller id.
maybe gmail is acting up on me i never saw this hit the list so im trying again. after finding out that chan_sccp is super buggy on my 12sp (ie hitting the wrong key crashes asterisk phone reboots , doesn't ring etc.) I went back to using chan_skinny this seems to be stable but the only thing that is not working that i need is caller id to be displayed on the phone when it rings. Anyone know if this works with chan_skinny ? any help would be great. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling all Users to check out bug 2655
http://bugs.digium.com/bug_view_page.php?bug_id=0002655 Can anyone comment on what is proper or not. Thanks, Brian ___ 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] Anyone using stanaphone? Having small problem
> Try adding your DDI number to the end of the register: > > register => NUMBER:[EMAIL PROTECTED]/NUMBER > Tried that, didn't do anythig useful, same problem exists.. Any other ideas? ___ 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] Anyone using stanaphone? Having small problem
Below my stana-part from sip.conf - works fine :-) Regards ! - register => number:[EMAIL PROTECTED]/stana3002 [stana3002] type=friend username=number fromuser=number secret=passwort context=default host=sip.stanaphone.com fromdomain=sip.stanaphone.com insecure=very caninvite=no canreinvite=no qualify=yes nat=no disallow=all allow=gsm Brian Weaver schrieb: Try adding your DDI number to the end of the register: register => NUMBER:[EMAIL PROTECTED]/NUMBER Tried that, didn't do anythig useful, same problem exists.. Any other ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Advice on OS Choice
On October 16, 2004 02:24 pm, Michael Giagnocavo wrote: > And thus, you've just sealed how the lawyers are going to treat this: > "Manufacturer X could have been more careful and reduced the chances of > this tragedy occurring. Now all we can do is seek punishment for the people > who contributed to the loss of life." > You believe walking in and saying "Our policy states..." is going to work? I don't know. You don't know. It's up to the jury. At any rate I do believe that this thread has shifted slightly; it was at first about how having the software open source would make things bad; now it's about how the lawywers would make open source bad. Open-sourcing the control software to a critical system isn't bad, it doesn't make it more likely that someone will screw with the system. Someone else has already made a point that the schematics, service drawings/notes and very likely algorithms are already provided to the people who service the equipment. The point's moot, IMO; litigation has a funny way of making things completely nonsensical. I think this has been proven in this thread. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dropping last digit of phone number
Hi all, I've recently installed and configured Asterisk. I'm having some problems with phone numbers which look like 1 021 123 4567 (1 for an outside line) and then the phone number. Asterisk will always drop off the last digit and dial 1021123456 instead. I thought this was a problem with my contexts however I've recently added a SIP phone and it's initial context is the same as the analogue phones that display this problem the SIP phone works fine. Any ideas where I should be looking? Regards Demian -- Systems Consultant Core Technology Ltd 021 446 282 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Advice on OS Choice
On October 16, 2004 04:49 pm, Joe Greco wrote: > As a manufacturer, you build things and sell them, and you can recommend > whatever policies you like, but after it leaves the shipping department, > you're out of luck as to being able to guarantee any of that. Then, as a manufacturer, you should not be liable for what some dickhead in a service department is doing to it. :-) Like I said in my last message, litigation has a way of making things nonsensical. > > Firmware that boots checks image (or critical parts of image) for > > tampering against stored checksum (checksum that gets updated when > > correct update procedure is followed) -- Putz away, the firmware will > > still bring you to a full stop because it detected a problem. > That's highly complex; even Sun agreed there was no practical way to do it. > With a closed source system, it wasn't considered a risk, and since > everything up to the point where we received control from the OS was at > least very difficult to putz with, it wasn't checked /prior/ to execution. > Verification of the loaded kernel image happened after it was loaded, and > was designed specifically to catch things like disk blocks going bad. I dunno -- crytographically sign the images and verify signature on boot. Hell even a field hard drive swap would work in this case. > Again, the black box approach has advantages. Could you maybe engineer > something to verify stuff at each and every step, just so you could go open > source? Sure, perhaps, but at additional cost for more flash, and > additional cost for more development, and bad things then happen if you > do a field swap on hard drives to fix a broken unit, etc., and really it > becomes impractical. See above. > That's nice in theory, but potentially pretty darn expensive. Nobody > seemed to think that it was worth the trouble, expense, etc., to get so > paranoid about it. That's what I don't understand -- they're sufficiently paranoid when it comes to providing source, but security through obscurity is good enough to get past the legal department. Curious, really. > > To upgrade you can install the CD or reimage > > the drive with the new image, but you have to also replace the vendor > > key. > And how do you do /that/? You now need to have a keyboard attached to the > system to enter and replace the key? physical cartridge or smartcard that was shipped with the updated firmware, and "signed off" by someone who has the access code to authorize the firmware update. I dunno. Cryptographic signature on the images with the CA being the company releasing the firmware is even easier. > The point is that's all software. If it's open to inspection and > recompilation, it's easily open to defeat. I can make an init system that > is very difficult to reverse-engineer, complete with interlocks with any > other items that get launched, such that NOTHING happens unless that > process is happy, but if that can be replaced by an init that doesn't give > a fsck, because someone commented out all the code and recompiled it, then > we have trouble. *sigh* -- this is why I am saying that the boot firmware needs to make these checks, not the stuff you can tinker with when you have the source. Bootloaders only boot the end software, they're usually not too complex and once done require little to no maintenance. Keep *that* black boxed. Put the interlocks *there* -- your core system is still open to many eyes and a lot of scrutiny. > So, yes, you /could/ design such a system, and if you've open sourced all > your software, then you probably /have/ to. I would go on to say that you should have those checks and balances in place whether it was open or not... Hell those DURN TERRAISTS might decide to put rogue firmware out to make all the nuclear medicine machinery go critical. Yes, this is getting silly. > We're talking specifically about the case where distributing the source > makes it trivial for someone to work around those correct checks and > balances. You can't work around a check and balance like that -- firmware doesn't like the signature, it don't start up the executable. Capiche? We're talking about open-sourcing the main software, not the ROM bootloader (for lack of a better word: BIOS). > No, I'm not worried about that. The specific case that was of concern was > what happens when someone from the hospital campus electronics shop tampers > with the system, something bad happens, and then the system is reloaded > with a non-tampered copy, because hospital policy would be to send a > defective device back to the shop? These devices don't have some kind of audit log in them? Jesus. > Trusted computing is always a difficult thing. At a certain point, you > need to draw the line. Because we had a closed source solution, we were > able to fairly safely assume that when we got handed off at init, we had > a system which was likely in a known state, and could verify the loaded
Re: [Asterisk-Users] Re: Advice on OS Choice
On October 16, 2004 05:05 pm, Matt Riddell wrote: > Joe, could we stop this now? It's obvious that if you go to a GPL > project and start slinging mud at the GPL, you are in the wrong place. > I would recommend that you head over to a Microsoft mailing list where > I'm sure you will find an abundance of fodder for your outdated > methodologies. Just my opinion: he's not slinging mud at the GPL, he's (trying) to give a scenario where open-source is a Bad Thing. I get the impression that he's rather happy with the GPL in general. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Advice on OS Choice
On October 17, 2004 11:34 pm, Andrew Kohlsmith wrote: > On October 16, 2004 02:24 pm, Michael Giagnocavo wrote: ?? wtf happened to my list threading? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323: forcing 20ms packetisation
Hi all I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant. Our supplier of call termination requires h323 the following: * The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99. I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling. But the 20ms packetisation has me beat. I have made a test call to the provider which did not work becase I was sending 30ms voice packets. SO my question does any one know now to force the correct voice packet size ? Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Petulant losers thread [Advice on OS Choice]
Can all parties concerned drop this thread or take it offline. Craig - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, October 18, 2004 11:51 AM Subject: Re: [Asterisk-Users] Re: Advice on OS Choice > On October 17, 2004 11:34 pm, Andrew Kohlsmith wrote: > > On October 16, 2004 02:24 pm, Michael Giagnocavo wrote: > > ?? wtf happened to my list threading? > > -A. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem In RTC Client When Used With Asterisk
Hi When I call from 1 RTC Client to another without Asterisk everything use to be fine but when asterisk is there as a Registrar a problem use to occur in many calls, Caller can hear the voice of the receiving side but the receiver cant be able hear the caller for about 5 to 10 seconds, conversation will become two way after 5 - 10 seconds but this problem is a big hurdle in proper establishment of a call Does anybody ever had this problem ? Any suggestions will be higly apreciated Thanx in Advance ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ 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] Asterisk dropping last digit of phone number
On Mon, 18 Oct 2004, Demian wrote: > I've recently installed and configured Asterisk. I'm having some > problems with phone numbers which look like 1 021 123 4567 > > (1 for an outside line) and then the phone number. Asterisk will always > drop off the last digit and dial 1021123456 instead. I thought this was > a problem with my contexts however I've recently added a SIP phone and > it's initial context is the same as the analogue phones that display > this problem the SIP phone works fine. Any ideas where I should be > looking? I'd start in extensions.conf.. double-count your X's (or N's) in the exten=> lines to make sure they match the number you're trying to dial. You didn't mention much detail about how the analogue calls get into your *, nor how calls get out. I guess it shouldn't matter much; they'll all get routed through extensions.conf regardless. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thailand
What does anyone know about signalling in Thailand? Are there any issues with using Digium T1 or FXO/FXS cards there? -- Jayson Vantuyl ___ 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] compiling cdr_mysql on AMD64 fedora core 2
I had simillar issues (not the same maybe) with Centos 3.3 X64. The first was becuase I had asterisk compile in /usr/src/asterisk-1.0.1 rather than /usr/src/asterisk. creating a symbolic link took the build process further but still failed. This time it was to do with the fact that it was looking for the mysql libs in /usr/lib/mysql whilst being x64 they were installed in /usr/lib64/mysql. Once again creating a symbolic link fixed that and I was able to compile clean. I hope this helps you diagnose the issue that you are having (my guess is that the error you are reporting is simmillar to the first error I had) Umar. On Sat, 2004-10-16 at 21:52, david winter wrote: > I got this error when installing cdr_mysql on an AMD64 running fedora > core 2. Anyone have ideas on what is wrong? > > > > gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes > -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o > format_mp3.o format_mp3.c > > gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes > -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -shared > -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o > layer3.o tabinit.o interface.o format_mp3.o > > /usr/bin/ld: common.o: relocation R_X86_64_32 can not be used when > making a shared object; recompile with -fPIC > > common.o: could not read symbols: Bad value > > collect2: ld returned 1 exit status > > make[1]: *** [format_mp3.so] Error 1 > > make[1]: Leaving directory > `/home/dwinter/src/asterisk-addons/format_mp3' > > make: *** [format_mp3/format_mp3.so] Error 2 > > [EMAIL PROTECTED] asterisk-addons]# > > > > __ > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem In RTC Client When Used With Asterisk
HI, I have used RTC with other SIP Proxies like SER and party sip and it works fine, never tested it with asterisk though. Basically Asterisk initiallly proxies RTP through itself and then sends reinvites to both endpoints to make RTP flow directly between the two gateways. Asterisk does have problems with the packetization perid values. It might be the case that the RTC endpoints use a different packetization period as compared to asterisk and it is only when the RTP goes direct, the endpoints start using the same packetization. Whatever the problem maybe, I would suggest capturing SIP and media packets on both server and client side and analyzing them. You can use ethereal (www.ethereal.com) for this purpose, it is an extremely useful opensource network analyzer. Hope this helps, Danish ___ 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] Unusual protocols
examples of things which I have actually been asked about. There are a number of protocols based in 2600Hz tones (most US) and 2280Hz tones (mostly Europe), which are probably still spread quite widely in low density point-to-point connections. If there is anything you need, please tell me about it. I want to build a picture of what might be worthwhile tackling. You probably won't go far wrong by looking at the support offered by www.aculab.com and trying to match it . Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 190 "Dial-Plan String" Settings
Hello James, There is nothing special with the Snom phones. The empty dialplan string is normal. You only have to specify the displayname, account, password and registrar. I think you have a mistake in your extensions.conf. Does it work with another (soft)phone? Regards, Joris On Oct 15, 2004, at 1:51 PM, James Bean wrote: I am having a problem with my new SNOM190 and my asterisk box. Incoming calls to the SNOM work perfectly, but when i dial-out I get a "Not Found: " on the SNOM display everytime I try, nothing shows up on the console of the asterisk box so its not even touching it. I have the latest 3.54 firmware on it and when I looked at the Line 1 setup for my asterisk box I released that in the SNOM phone there is nothing in my "Dial-Plan String" I take it it matches this inside the phone to choose which line to use in the SNOM phone. Unfortunately I am not finding much on the format of the Dial-Plan String in the SNOM phones. All I need is for it to send all calls regardless of format to the asterisk box. Anyone got any suggestions. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk AGI 'Get Data' escape digits not working on long records
Hoping someone can please help me. I have written an AGI application (that uses the Asterisk-AGI perl library) that processes requests to record wav files, capture dtmf, return dtmf etc to my dial plan. It works well, except when I record a long recording ( I have not been able to figure out a direct pattern - but approximately 40 minutes or longer of total recording in MSGSM format) It will no longer respond to my DTMF escape digits. In my agi-test.agi file I simply something similar to the following. $result = $AGI->record_file($wavfile, WAV, 12345 , 7, 1); As expected it will wait for up to 1 digit and return the value in ASCII into $result HOWEVER I need it to sometimes record up to a maximum of 3 hours. (108 ms) $result = $AGI->record_file($wavfile, WAV, 12345 , 108, 1); But it gets to maybe more than half an hour, is still recording fine but NO MATTER WHAT digits i press, it never escapes from this command when i constantly try pressing any of the escape digits. Does anyone have an insight or similar issue? I wish i could resolve this one, it is killing me. Thanks Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cross-connecting dynamic channels
Hi, is it possible to cross-connect dynamic channels? I was trying to do someting like this in zaptel.conf: #first interface dynamic = eth,eth1/00:40:F4:A4:7C:5C,24,2 bchan=1-23 dchan=24 #second interface dynamic = eth,eth0/00:40:F4:A4:7D:FE,24,2 bchan=25-47 dchan=48 dacs=1-24:25 but ztcfg is always giving me back something like: line 160: Channel 1 already configured as 'Individual Clear channel' at line 149 ... line 160: Channel 24 already configured as 'D-channel' at line 150 Can something like this be done, and if so, how should i configure the channels? thanks a lot katharina -- GMX ProMail mit bestem Virenschutz http://www.gmx.net/de/go/mail +++ Empfehlung der Redaktion +++ Internet Professionell 10/04 +++ ___ 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] Asked to transmit frame type 64, while native formats is 8
Think i solved that one by the ordering of allow= in sip.conf At 18:39 2004-10-17, you wrote: http://bugs.digium.com/bug_view_page.php?bug_id=0002519 If anyone has seen that error please come forward and report on this bug please. The original reporter is unwilling or unmotivated to even make an effort to assist in correcting the issue. So if anyone else has seen this please post. Thanks, Brian ___ 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
