Re: [Asterisk-Users] ATA's
On Tue, Feb 15, 2005 at 01:40:24PM -0600, Matthew Boehm wrote: > > wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is > > a "bridge mode", where the LAN and WAN ports would act just like a > > switch, so that you can easily chain devices without routing/NAT. Just > > like most IP phones do. > > So what happens if you try and chain a bunch of SPA-2100s? Does the 2100 > act more like a router? > The SPA-2100 will perform NAT between the LAN and WAN ports. It means that you won't be able to connect to devices "behind" SPA-2100 from the WAN side (except by configuring DMZ, but it's gets awful then). It also means that you can't chain them with their basic config, because it won't like having the same network address (192.168.1.x) on both its interfaces. So right now, if you want to chain them, you have to play with IP addresses, DHCP settings, etc. Not fun, particularly when you consider that this device is really "plug and play", it remotes configure everything. Hopefully a "bridge" mode will appear in a later firmware upgrade (which, for Sipuras, are frequent and readily available on their website). -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
On Mon, Feb 14, 2005 at 10:47:23PM +0900, Hermann Wecke wrote: > Matthew Boehm wrote: > >[...] In the meantime, get a Sipura 2100, supports 2 729 calls and > >has both WAN/LAN ports. > > I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying > one to test. Street price around US$ 90. > Another one with dual g729 channels is MTA V102. Street price US$ 100. > Also will test this one. > > I'm still looking for other units with dual g729 channels... > Back in december, the Uniden "was supposed to do 2xG729 at a later time". Not sure if the current firmware allows it. BTW, I've been fairly disappointed with Uniden firmware and their release cycle : their hardware is great, but they take months to release new firmwares, even when "phone crashing" bugs are discovered. If you want 2xG.729 now, working reliably, for under $90, you can't go wrong withe the SPA-2100. The only thing the SPA-2100 (still) lacks is a "bridge mode", where the LAN and WAN ports would act just like a switch, so that you can easily chain devices without routing/NAT. Just like most IP phones do. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
On Sun, Feb 13, 2005 at 10:39:36AM -0800, Luki wrote: > The Sipuras have a ton of configurable parameters. If you understand > them (and there is no good manual, unfortunately) then you can be of > great benefit. Otherwise they'll be worthless. I particularly miss the > dial-plan, distinctive ring and audio gain options on the > Grandstreams. Remote syslog can also be useful for debugging. It all > depends what you need, I guess. > > Further, the Sipuras have a more detailed status, that is accessible > WHILE you are engaged in a conversation. > > I think you're paying a bit more for the 1000 (1 line version) as > compared to the Grandstream 286, but if you need/want two independent > lines, then the Spa 2000 is more economical (as Peter said). > The Sipuras are really a dream to manage, particularly in an international environment. You can customize the tones, the rings, the voltages, the dialplan, the features... well, everything. They are (securely) remote manageable and upgradeable. They are rock solid. Sipura support is helpful in case you need them for complex issues. Voice quality is top notch. The Grandstreams are less manageable, have less parameters, have only american tones, no dialplan support, no auto-upgrade (well, they recently added some kind of support). Voice quality is OK. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
On Sun, Feb 13, 2005 at 07:43:06PM -0600, Matthew Boehm wrote: > We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN > port. Only downside is that only 1 call can be using 729 at a time. This has > been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to > overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and > has both WAN/LAN ports. Personally, I dislike the lack of LEDs on the 2100. > My 2100s have 3 LEDs, plus 2 for each RJ-45 port. Instead of just 2 for the SPA-2000. -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Sipura SPA-841 SIP phones
On Thu, Feb 10, 2005 at 09:07:58AM -0500, Giovanni Powell wrote: > Nothing to do with your question, but by any chance, when you plugged > the phone into the wall did you hear a dialtone or is this something > generated by asterisk On a SIP phone, the dial tone is locally generated. The Sipura will only generate a dial tone if registrered. BTW, you can easily check on the Sipura web interface that the dial tones are parametered there. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-quality, high-bandwidth codecs?
On Tue, Feb 08, 2005 at 02:58:01PM +0100, Roy Sigurd Karlsbakk wrote: > >>are there any codecs around that allows high quality as in "studio > >>lite"? it may consume high bandwidth, and hopefully allow some packet > >>loss. > >> > > > >I'm not sure what "studio lite" means to you. Maybe hard figures would > >be more precise. > > > >G.722 might be interesting : 64 kbps, 7 kHz. It's not free. > > > >Otherwise, MP3 or OGG might be ok ? > > Would it be hard to do a codec_ogg? > It would rather be a codec_vorbis, as Steve pointed out. It's definetly feasible. However, I'm not sure how useful it would be. You'd need some kind of device talking Vorbis to Asterisk. Does it exist ? -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high-quality, high-bandwidth codecs?
On Tue, Feb 08, 2005 at 12:26:39PM +0100, Roy Sigurd Karlsbakk wrote: > hi > > are there any codecs around that allows high quality as in "studio > lite"? it may consume high bandwidth, and hopefully allow some packet > loss. > I'm not sure what "studio lite" means to you. Maybe hard figures would be more precise. G.722 might be interesting : 64 kbps, 7 kHz. It's not free. Otherwise, MP3 or OGG might be ok ? -- Nicolas Bougues ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE or IAX?
On Sun, Dec 19, 2004 at 06:56:19PM +1100, Eric Bishop wrote: > Hi all, > > Information on this topic seems a little scarce, so I thought I'd try > the list > > Apart from the the coolness factor can anyone explain to me in what > situation one would use TDMoE rather than IAX for communication > betwwen 2 Asterisk servers? > I thing that you're mostly better with IAX between 2 Asterisk servers. TDMoE, however, is not limited to Asterisk. It's part of zaptel. You can use it to transport a TDM link over an Ethernet network (or IP, with some kind of tunneling), and get it back as a TDM link on the other side (with proper hardware). -- Nicolas Bougues Axialys Interactive ___ 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] UPDATE - Echo cancellation, when software doesn't cut it. Whats next?
> > So the two questions remain. > > 1. Why do incoming calls have nearly no echo (sound great), and outgoing > calls are bad during the first 30 seconds, and okay (but not good) after > that. > > 2. Why do outgoing calls to cell phone numbers sound great? > > Seeing as an outgoing call to a land line has echo, but the same land > line calling in has virtually no echo, does this point the finger at > Asterisk code having issues? > Echo (most often) comes from hybrid circuits on PSTN lines (2 wires <-> 4 wires transformation). Cell phones, as well as some corporate digital phones don't go through that kind of devices, so there is no echo generated. So, basically, no echo cancellation required. Unfortunatly, it's impossible to know from the caller point of view whether the call will need echo cancellation or not. -- Nicolas Bougues Axialys Interactive ___ 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] VoIP hackers gut Caller ID
On Wed, Jul 07, 2004 at 11:57:31AM -0400, Timothy R. McKee wrote: > This has always been one of my pet peeves, even as I worked in the industry. > A telco switch operating a DS1 on trunk side should enforce caller-id > numbers to be within the range of DID numbers assigned to that trunk. There > should be a default DID number that is used to replace any *invalid* numbers > sent on that trunk. Note that blocked caller ids would still be blocked, > but the rest of the data should be corrected. Blocking ID is ok, lying > about it is not. > > Blind trust of a non-SS7 link is a _bad_ thing. > PRI signalling enables "Network provided" or "User provided" caller-id. Maybe IAX could implement such a thing. It's very common in France (at least) : - the network will provided a guaranteed caller-id - the user (CPE) may provide another one (usually, a DID number) and the called party gets both. Unfortunatly, as far as I understand, Asterisk is not really designed to handle more than one caller id number. -- Nicolas Bougues Axialys Interactive ___ 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] Multiple E1s over TDMoE?
On Sat, Jul 03, 2004 at 07:44:54PM +0200, Thilo Salmon wrote:
>
> How would you go about running, 8 or 16 say, E1s over TDMoE? Would you
> create multiple dynamic spans or just one large one? How would you
> assign d channels to spans, if you had just one large span?
>
> Did any of you guys try this before?
>
Somewhat, yes.
I run 4 TDMoE E1s between pairs of servers (4 E1s between each of
them).
What I think about that :
- it works
- TDMoE doesn't like SMP. It doesn't like running on a NIC used for
other kind of traffic. It will crash your box under heavy (non
TDMoE) load. I believe that there must be some race condition
related to dev_queue_xmit(), which is probably not callable at
anytime.
- the subaddr support is not complete in the released driver. Here is
a patch that will handle it (as described in zaptel.conf)
- it is not 100% reliable. You will get frame drops, and you will
notice it if you look at your D-channel dumps.
- use high quality NICs and switches.
--
Nicolas Bougues
Axialys Interactive
--- ztd-eth.c.old 2004-02-01 06:53:58.0 +0100
+++ ztd-eth.c 2004-07-11 00:51:45.0 +0200
@@ -251,7 +251,7 @@
{
struct ztdeth *z;
char src[256];
- char tmp[256], *tmp2, *tmp3;
+ char tmp[256], *tmp2, *tmp3, *tmp4 = NULL;
int res,x;
unsigned long flags;
@@ -273,6 +273,7 @@
return NULL;
}
if (tmp2) {
+ tmp4 = strchr(tmp2+1, '/') +1;
/* We don't have SSCANF :( Gotta do this the hard way */
tmp3 = strchr(tmp2, ':');
for (x=0;x<6;x++) {
@@ -288,7 +289,8 @@
} else
break;
if ((tmp2 = tmp3))
- tmp3 = strchr(tmp2, ':');
+ if (!(tmp3 = strchr (tmp2, ':')))
+ tmp3 = strchr (tmp2, '/');
}
if (x != 6) {
printk("TDMoE: Invalid MAC address in: %s\n", addr);
@@ -300,6 +302,25 @@
kfree(z);
return NULL;
}
+ if (tmp4) {
+ int sub = 0;
+ int mul = 1;
+
+ // We have a subaddr
+ tmp3 = tmp4 + strlen (tmp4) - 1;
+ while (tmp3 >= tmp4) {
+ if (*tmp3 >= '0' && *tmp3 <= '9') {
+ sub += (*tmp3 - '0') * mul;
+ } else {
+ printk("TDMoE: Invalid subaddress\n");
+ kfree(z);
+ return NULL;
+ }
+ mul *= 10;
+ tmp3--;
+ }
+ z->subaddr = htons(sub);
+ }
z->dev = dev_get_by_name(z->ethdev);
if (!z->dev) {
printk("TDMoE: Invalid device '%s'\n", z->ethdev);
@@ -311,7 +332,7 @@
for (x=0;x<5;x++)
sprintf(src + strlen(src), "%02x:", z->dev->dev_addr[x]);
sprintf(src + strlen(src), "%02x", z->dev->dev_addr[5]);
- printk("TDMoE: Added new interface for %s at %s (addr=%s, src=%s)\n",
span->name, z->dev->name, addr, src);
+ printk("TDMoE: Added new interface for %s at %s (addr=%s, src=%s,
subaddr(net byte order)=%d)\n", span->name, z->dev->name, addr, src, z->subaddr);
spin_lock_irqsave(&zlock, flags);
z->next = zdevs;
@@ -350,3 +371,6 @@
module_init(ztdeth_init);
module_exit(ztdeth_exit);
Re: [Asterisk-Users] Zaptel dacs / dacs
On Fri, Jul 02, 2004 at 02:48:17PM -0700, Chris A. Icide wrote: > > 2) can you cross connect PRI interfaces? > > in other words can you use the dacs functionality to insert a digium card > (on a system running asterisk) in between a pri from a carrier, to a legacy > pbx system? > Yes, you can. But what's the point ? With such a setup (zaptel bridging), you don't get that many benefits. I developed a small std-local driver that can help you : - cross connect two PRIs - and get a monitoring feed (imagine a "Y" cable), so that you can do (read-only) analysis on the PRI traffic. But I doubt that's what you want to do there. What you may want to do is being able to use the legacy PBX, as well as Asterisk features. For that, you need to : - setup Asterisk for the PRI connection to the telco (pri_cpe) - setup Asterisk for the PRI connection to the legacy PBX (pri_net) - setup your dialplan (extensions.conf) so that : - Asterisk talks to the telco the right way - Asterisk talks to the PBX the right way - Asterisk forwards calls to/from the PBX the right way - Asterisk does whatever else you want it to do It's not that hard. However, it will probably involve some "trial and error", and your PBX users might not like it. -- Nicolas Bougues Axialys Interactive ___ 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] Digital Line Distortion
On Mon, May 03, 2004 at 09:28:56PM +1000, Adam Goryachev wrote: > Firstly, the problem... > > Ever since I installed and setup asterisk, I have had various problems, > initially it was echo caused by the ISDN (isdn4linux) card I was using. > So, I upgraded to the X101P from digium. I still had echo, so I figured > it was also caused by the ATA186 (cisco) I was using. So, I upgraded > again to the TDM40B quad FXS card. This solved pretty much all my > problems, except eventually, I needed more incoming lines. So, again, I > upgraded to a digital line (10 channel PRI/E1) and purchased the brand > new TE405p from digium... Now, eventually I got this working properly, > for incoming and outbound calls, I have incoming callerid working, > etc... > > However, ever since I did this, I continually get complaints from people > about how terrible my phone lines are. Not *everyone* complains, but > most people do > We did face what may be the same problem here. The problem came from the fact that on some motherboards (well, *most* motherboards, as far as I tested), the TE405P has a problem which makes it send every one in 8 (or was it 16?) bytes as 0xFF (instead of whatever the U/A-law value may have been). On the RX side of things, it was always perfect, thus when connecting to a local IP phone we heard a nice sound, but the remote party always had a quite garbled output. You can check it quite easily : - plug a crossover cable between two ports - do not start Asterisk (but load and ztcfg everything) - cat /dev/zap/span1/1 on one terminal - ls >/dev/zap/span2/1 on another terminal (provided that spans 1 and 2 are connected together) - if everything works well, you should have a perfect output for your ls on the "cat" terminal. Otherwise, try hexdump and watch the columns with FF. It was solved by using a PCI 2.2 compliant motherboard (i865 based). It's quite an odd behaviour, and it's still not clear to me why it happens. I initialiy thought it could be solved by fixing the FPGA VHDL, but I'm not an expert in that field. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream 1.0.4.55 Firmware
On Fri, Apr 30, 2004 at 06:13:49PM +0100, Senad Jordanovic wrote: > [EMAIL PROTECTED] wrote: > > Go to 1.04.54. This is pretty stable. Find it at > > www.telappliant.com/grandstream > > > Does this version supports TFTP auto configuration? If it does, please > contact me off the list for volume purchase discussion! > Err, all (1.0.4.x at least) GS firmwares support TFTP autoconfiguration ! -- Nicolas Bougues Axialys Interactive ___ 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] Best echo-free and trouble-free system?
On Fri, Apr 30, 2004 at 10:28:02AM -0500, Barton Hodges wrote: > [EMAIL PROTECTED] wrote: > > The real problem arises when : > > - you have some echo induced somewhere (your call goes through a 2 > > wire line) > > - you have some delay induced somewhere (you use VoIP for instance) > > Following the "2-wire to 4-wire causes echo" thought, the following > should not result in noticable echo, true? > > - Analog phone <-> TDM10B-FXS <-> Asterisk <-> TDM01B-FXO <-> PSTN Yes, because although echo will exist, delay should be short enough so that you don't notice. Never tried such a setup myself, though. Please furthermore note that Asterisk uses "pseudo" TDM. In real telco world, PCM highways that interconnect trunks and devices switch one byte every 8000th/sec. OTOH, Zaptel devices switch eight bytes every 1000th/sec. This is due the to PC bus architecture (it would cause way too much overhead otherwise). So the delay is actually 8 times longuer (at least) than in the PSTN. > - VOIP Phone <-> Asterisk <-> VOIP Phone > - VOIP Phone <-> Asterisk <-> T100P <-> PRI > Nobody's supposed to generate echo on VoIP phones. However, the "PRI" side will probably connect to a 2-wire PSTN set at the remote end, so you will get echo from there. > However, the following could result in noticable echo (as I am > experiencing): > > - Analog phone <-> ATA <-> Asterisk <-> TDM01B-FXO <-> PSTN > - VOIP Phone <-> Asterisk <-> PSTN > Definetly. Although Asterisk (zaptel, actually) make a fairly good job at cancelling it. > What about the following as described in Raymond McKay's setup (Thank > you Raymond) > Does the channel bank provide the needed, and adequate echo > cancellation? > I don't have any experience with channels banks. Not very common stuff in Europe. > > Since you state that echo cancellation needs to be performed closest > to the source, could the Grandstream HandyTone-286 be doing an > inadequate job of echo cancellation? If this is the case, does anyone > have experience with another ATA (Sipura SPA-1000, Cisco ATA-186, > etc.) that does such a great job of echo cancellation, that the > "2-wire to 4-wire" situation is not an issue? Does Grandstream have > improved echo cancellation scheduled for a future firmware upgrade? > The Sipuras are definetly better than the HandyTones. I've heard that the forthcoming GS firmwares will enhance echo cancellation performance, though. -- Nicolas Bougues Axialys Interactive ___ 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] MWI forwarding
As far as I understand how voicemail is integrated into Asterisk, it seems that SIP channels poll MWI directly from the filesystem. Is it possible (feasible?) to have something like : - a central voicemail server - which has an IAX peer with a mailbox= line with tens of VM boxes - this peer has itself tens of SIP phones connected to it. And it would alert the SIP phones when it receives MWI over the IAX channel from the central server. -- Nicolas Bougues Axialys Interactive ___ 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] Help choosing a UK IAX provider
On Wed, Apr 21, 2004 at 04:18:51PM +0100, Craig Waddington wrote: > > 1 1 ms 2 ms 1 ms 10.5.0.1 > 217 ms14 ms14 ms 195.10.119.94 > 317 ms14 ms14 ms 195.10.119.158 > 422 ms14 ms15 ms 217.23.160.1 > 515 ms15 ms31 ms 217.23.162.122 > 617 ms15 ms14 ms 217.23.160.85 > 719 ms18 ms14 ms 217.23.160.186 > 830 ms26 ms29 ms tier1-1.BUD2.psie.net [154.14.68.113] > 931 ms39 ms29 ms linx1.teleglobe.net [195.66.224.51] > 1026 ms28 ms30 ms if-0-0-0.bb2.London.Teleglobe.net > [195.219.96.81 > ] > 1159 ms87 ms 108 ms ix-3-1-0-822.bb2.London.Teleglobe.net > [195.219.2 > .34] > 1276 ms54 ms54 ms wi2.westloc.com [82.145.32.2] > 13 229 ms 239 ms 187 ms wc3-10.westloc.com [82.145.32.73] > This last hop may be the source of your problem. Since I believe it's not a trans-continent link, it's either : - a very congestioned link - a router with serious problems at hop 13 (or maybe 12). You should contact whoever manages westloc.com -- Nicolas Bougues Axialys Interactive ___ 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] TE410P zaptel Driver Situation
On Wed, Apr 21, 2004 at 01:17:23PM -0700, Scott Stingel wrote: > > > >Dear Scott > the "reject" warning is a bug? I must put in bug track? > >Thanks in advance > >Dimitri > > No, not a bug I don't think. A warning that the framer driver was not able > to keep up with the PRI bit stream. > Zaptel hardware handles HDLC framing in software. Thus, if for some reason a Zaptel frame (8 bytes) is lost, a full HDLC frame will be lost, and it will be detected on the next one, thus the error message. The main reason it can happen is probably because of the server load. The driver can't poll the zaptel device fast enough. However, it's not clear to me how it can happen on busmaster devices such as T405P and T410P. Maybe a double buffering issue, or PCI bus load. Or maybe an interrupt fault; the Zaptel hardware provides a kHz interrupt to the driver for polling, and an interrupt might get lost (particularly if the IRQ line is shared). However, it seems to be detected by the driver (and it should print a warning). -- Nicolas Bougues Axialys Interactive ___ 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] Hot plug PCI?
On Tue, Mar 30, 2004 at 10:27:28AM -0600, Charlie Hedlin wrote: > The quick question: Do the digium drivers for the Digium Wildcard TE410P > (4 port T1/E1/PRI 3.3v card) , the T100P (single port T1), and the > TDM400P support hot plug PCI? I am also noting that while the TDM400P > doesn't state the voltage requirements, it looks like a 5v card. I hope > that I am wrong on this. > Hotplug PCI has two sides of things : - drivers and software must support hot plugging : there is support in the Linux kernel for that. Digium drivers don't support that yet, but this is not very hard to do. - hardware : you must have hotplug capable boards and bus. I believe the only option there is CompactPCI, and of course cPCI boards and chassis are very different from your day-to-day PCI stuff. Of course Digium boards, as well as any other standard PCI board does not fit. cPCI boards and chassis are considerably more expensive than their standard counterparts, if they exist at all. You can't just hot (un)plug a standard PCI board : the bus is not meant for this, you have most chances of destroying your board and/or your bus. -- Nicolas Bougues Axialys Interactive ___ 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] High latency from Europe, 500-800ms.
On Fri, Mar 19, 2004 at 08:38:49PM +0200, [EMAIL PROTECTED] wrote: > We're using a 7940 from Europe, connecting to a US Asterisk server, and > it works great. We setup a local Asterisk server in Europe, had the > 7940 connect to it, and used IAX2/GSM to connect to the US. It is > choppy using all CODECS, and I am curious if there are any > recommendations on getting this to work well? I'd rather not have the > phones connect directly to the US. > What I can tell you is that we are located in Europe, and have several users in Canada, often using G.711 as codec, and the communication is almost always cristal clear. So your problem is probably related to network congestion somewhere on your side, on the remote side or in transit. traceroute is your friend. -- Nicolas Bougues Axialys Interactive ___ 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] Identifying a call with manager interface
On Fri, Mar 19, 2004 at 10:10:57AM -0500, mattf wrote: > You can also use a unique CallerID, that's how I got call tracking to work > with my Asterisk Central Queue System(backend part of the astguiclient). > Take a look at the code if you like: > That's what I thought about. However in this case caller id is supposed to be "right", at least for its numerical part. I'll try to use the name part of the caller id for that purpose. -- Nicolas Bougues Axialys Interactive ___ 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] Identifying a call with manager interface
On Fri, Mar 19, 2004 at 03:58:12PM +0100, michiel betel wrote: > > You should be able to specify an ActionID with the originate request. > Asterisk will the put this ActionID in all replys to your request. > Haven't (yet) tried this myself, but check manager.c for the exact > implementation. > ActionID is specified in the Response message. It would be useful with interleaved Requests/Responses, which is not the case here. However, the ActionID is not specified in the "Event" messages I get meanwhile, so I can't track call progress nor channel allocation. -- Nicolas Bougues Axialys Interactive ___ 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] Identifying a call with manager interface
Dear all, I'm trying to play with the manager interface. What I'd like to do is being able to originate a call and trace its status through events. I use the "Originate" manager command. I then receive several events telling me about the progress of the call, and then the "Response" message. However, I didn't find a way to be sure that the first "Event" I receive after the "Originate" really relates the call I'm making, and not some other random call, since I believe that I may get events for any channel, not just mine. Note that the Channel I'm using is IAX based, and looks like this : IAX2[217.146.224.41:4569]/3 in the events messages. So I have no way to know it's really mine. Event the final Response message doesn't state the "UniqueId" of the call. Maybe I missed something obvious. Any idea ? -- Nicolas Bougues Axialys Interactive ___ 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] Does anyone have faxes working well with X100P and TDM40B cards?
On Tue, Mar 16, 2004 at 02:16:01PM -0600, Jim Sneeringer wrote: > > Why does Asterisk not work as well with fax modems as any other phone > system? > Because Asterisk does TDM in software. It's handled by 64 bits chunks, instead of bit by bit on TDM hardware. It has two consequences : - if for some reason (timing, load, bad luck...) a chunk is lost, you loose 1 ms of sound. This matters for modems. - handling sound 1ms at a time induces some slight delay (>1ms), but which can easily produce echo on analog loops (you don't hear it when it's 1/8000th of a second later, but if can matter if it comes back a few ms later). Thus Asterisk uses echo cancellation, which may alter data transmission. Hopefully tone detection deactivates it though. -- Nicolas Bougues Axialys Interactive ___ 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] about voice conference system. I need suggests
On Tue, Mar 16, 2004 at 10:10:01PM +0800, wangji wrote: > >I am trying to deploy a conference system. There are 16 fxs ports, 16 fxo > ports and 32 conference resource(device) with conference management system. > It should support max to 16 parts in a conference and max to 8 conferences. > And I only have 1 week to do that. > Looks ambitious... >I want to use Dialogic boards: 1 D160sc + 1 MSC160sc + 1 DCB320sc. I > choice Dialogic because of I have wrote codes on Dialogic hardware. I have > wrote codes use Globecall under Dialogic Linux release 5.1. And I never > found those hardware like Quicknet, Zaptel, LineJack, Pika, VoiceTronix, > etc. in my country. These hardware should spend my time to choice, buy and > study. In my country, people usually use Dialogic, NMS and audiocodes. > Zaptel devices don't have widespread retail channels. But I believe you can buy them from about anywhere on earth, provided that you can do a wire transfer and receive the Fedex guy. It's certainly possible in China. >I find three open source softwares from Google: GNU Bayonne, GNUCOMM and > Asterisk and other way like PublicVoiceXML. I read the documents of those > softwares. It seems to not support Dialogic conference board DBC-320. I > don't know if one of them support at this time. Who know that? > Asterisk is designed to use the host CPU to perform various kind of tasks (signalling, signal processing including conferencing, etc.) that are performed by embedded processors on Dialogic hardware. Thus, it makes no sense to have support in Asterisk for every kind of Dialogic board : they are overkill, and overpriced, to be used with Asterisk no matter what. >I need to choice a solution to finish that job. It should be easy, > powerful enough, cheaper, convenient. Give me some suggests about software > and hardware and other things. Any help is appreciated. > There are various non-free software products that can handle this stuff. Your Dialogic dealer is probably able to get you in touch with a Dialogic solution provider. It seems to be what you're looking for (see below). >By the way, anyone can tell me the commercial license price of these GPL > software: GNU Bayonne, GNUCOMM and Asterisk. > I believe no "commercial licence" exist for GNU . For Asterisk, you should contact Digium directly (but then again, you should use zaptel hardware). -- Nicolas Bougues Axialys Interactive ___ 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] Fax redirection problem
On Fri, Mar 12, 2004 at 11:38:53AM -0600, Tilghman Lesher wrote: > On Friday 12 March 2004 05:41, Nicolas Bougues wrote: > > From time to time, during a conversation, Asterisk seems to detect > > a fax tone. > > > > It then tries to redirect it, and prints the following message : > > > > Redirecting Zap/2-1 to fax extension > > > > According to the source, it does this only if it matches a "fax" > > extension in the current context. > > > > I don't have a "fax" extension, but a wildcard one (_.). I would > > like these detections to be simply ignored. Is there any way to do > > it ? > > Change your wilcard extension to "_X.". That will require all of your > extensions to start with a digit. > Thanks for the tip :) It's ok now. -- Nicolas Bougues Axialys Interactive ___ 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] European Caller ID
On Sun, Mar 14, 2004 at 11:45:38AM +0100, randulo wrote: > > Here in France, we currently do not have it enabled on any of out lines, > but I would consider paying for it if I thought * via X100p cards would > be able to detect it properly. > Can't say, I have no such board. > I know for example that when people are calling us from the US or from > London CID is not captured by France Télécom, because we have a free > service here where you can call a number to see if you missed any calls. > FT does forward caller id when it has a chance to. But foreign carriers may use strange "path" in call routing, and may not be able to be able to send proper caller id. Basically, you will only see it if the foreign caller has direct SS7 connectivity. Otherwise, the foreign carrier is probably forwarding the call via some PRI, and often can't hide himself without hiding the original caller as well. > Are the technical "specs" of CID different all over? Comments, > experiences please? CID, at least in France, exists in two flavours : - carrier guaranteed, called NDI in France - user provided (and thus, not guaranteed by the carrier), called NDS in France On ISDN, the (french) carrier will usually provide both (out of which Asterisk will usually use the NDI). On POTS, France Telecom will usually forward the NDS if it looks correct (9 digits without leading 0), the NDI otherwise. International CID is usually carried in E.164 format (and may, or not, be dialable, depending on your local private exchange configuration). -- Nicolas Bougues Axialys Interactive ___ 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] PCI front mount chassis?
On Fri, Mar 12, 2004 at 08:41:36AM -0500, Walt Reed wrote: > > The voice cards generate an order of magnitude more interrupts than > anything else. This "may" be why it's not recommended to share > interrupts on voice cards. Don't know if the T1 cards have a similar > issue. I would hope not. The x100p's are a pretty simplistic device. > They probably generate an interrupt for every byte. The x100p's are also > used for timing in things like MOH and MM conferences AFAIK. It seems > like it would be nice to only put one card in "timer mode" if that is > indeed what is generating all those interrupts. Could someone "in the > know" enlighten us? > Digium boards usually generate 1000 interrupts/sec. This is mostly a "timer" interrupt, so that the driver can poll the board for 8 bytes per channel 1000 times per second. The polling is either slave (PIO), or busmaster (DMA). Each board generates 1000 interrupts/second, no matter the kind/number of ports on the boards. The driver knows if that's a single channel board, with 8 bytes to fetch, or quad E1, with 128*8 bytes to get on each interrupt. There are quite strict timing requirements : if one interrupt is lost, 1/1000th of sound on the line(s) is lost, which can be quite bad for things like HDLC (on T1/E1) or modem sound (no matter the kind of channel). Sharing an IRQ may introduce some latency, because the IRQ has to be handled by several drivers, and if each of them does not behave properly, frames can be late/lost. -- Nicolas Bougues Axialys Interactive ___ 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 problem
>From time to time, during a conversation, Asterisk seems to detect a fax tone. It then tries to redirect it, and prints the following message : Redirecting Zap/2-1 to fax extension According to the source, it does this only if it matches a "fax" extension in the current context. I don't have a "fax" extension, but a wildcard one (_.). I would like these detections to be simply ignored. Is there any way to do it ? -- Nicolas Bougues Axialys Interactive ___ 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] PCI front mount chassis?
On Fri, Mar 12, 2004 at 06:48:41AM +, Jeff Stohl wrote: > I am running six reliably right now. > > Surely I am not the only one doing a large capacity single site? > You mean 6 quad span boards ?? In the T1 world, it's 576 channels on a single PC, 720 if E1. What kind of configuration can handle that load reliably ? What's the application ? -- Nicolas Bougues Axialys Interactive ___ 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] PRI errors blocking Asterisk
On Thu, Mar 11, 2004 at 08:13:48PM -, Scott Stingel wrote: > could you please post your zaptel.conf? > Here it is : span=1,1,0,ccs,hdb3 span=2,0,0,ccs,hdb3 span=3,1,0,ccs,hdb3 # Colt est source de timing span=4,0,0,ccs,hdb3 defaultzone=fr bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 -- Nicolas Bougues Axialys Interactive ___ 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] PRI errors blocking Asterisk
On Thu, Mar 11, 2004 at 04:48:42PM -, Scott Stingel wrote: > Hi Nichoas- > > Are you are getting lots of frame re-transmission messages in > /var/log/asterisk/messages as well? > No. I get a few of these messages, though : Mar 11 16:11:11 WARNING[81926]: PRI: Read on 131 failed: Unknown error 500 Mar 11 16:11:11 NOTICE[81926]: PRI got event: 8 on span 2 Mar 11 16:11:11 WARNING[81926]: PRI: Read on 131 failed: Unknown error 500 Mar 11 16:11:11 NOTICE[81926]: PRI got event: 8 on span 2 Note that I'm not sure about the timing settings on my board. The board has (currently) 3 E1 spans connected, from 3 different operators, all of them providing a clock (no guarantee they are synchronised). Upon module loading, the driver says TE410P: Timing from source 0 I chose one (quite random) E1 span as a primary sync source in zaptel.conf SPAN 3: Primary Sync Source I'm not sure how this setting is used. Do I really have to set one ? -- Nicolas Bougues Axialys Interactive ___ 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] PRI errors blocking Asterisk
On Thu, Mar 11, 2004 at 05:12:24PM +0100, Klaus-Peter Junghanns wrote: > Nicolas, > > does your TE405P share the irq? No, it's alone on IRQ 17 (with IO-APIC). -- Nicolas Bougues Axialys Interactive ___ 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] PRI errors blocking Asterisk
Hi Asterisk community, Every once in a while (can be several times a day, or every few days), I get that kind of error (with a TE405P) : PRI: Short write: -1/66 (Unknown error 500) After that, the E1 links on the server get jammed : all the current channels, or any new zap channel is simply unkillable. Restarting Asterisk (after kill -9) solves the problem. It seems to me that the Q921 layer in libpri has an unrecoverable error (such as the fd being wrong/closed). Anybody know where it could come from, and/or what should be done to avoid it ? -- Nicolas Bougues Axialys Interactive ___ 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] G.729 vs. G.729 pass thru
On Wed, Mar 10, 2004 at 04:57:37PM -0500, Derek Samford wrote: > Anthony, > Asterisk by default allows pass through. You shouldn't need a > license. It's only when you need to do transcoding (I.E. you need to > decompress the voice, whether it be for Codec translation, to dial out a > Zap channel, which would be ULAW or ALAW, Voicemail (still technically > codec translation), or any other number of things.) If you are just > doing protocol conversion, with the same codecs, then your fine. > However, if you need > SIP(any codec other than G.729) -->asterisk ---> H.323 SS(g729) > Then you would need a license. > I have a similar question : imagine you have G729 licences, and Asterisk performs "native bridges". Asterisk doesn't do any conversion in this case. Are G729 licenses counted as "in use" ? -- Nicolas Bougues Axialys Interactive ___ 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 Codecs [G.729]
On Wed, Mar 10, 2004 at 09:03:57AM -0800, wrote: > Rich, > > In real world, using real tool, getting real number. You don't expect to > either talk only mode or listen only mode. Per call must have Rx & Tx for > inbound & outbound. > [...] > > Engineering rate is per channel but to calculate the bandwidth consumsion, > it must be in realtime full-duplex. > Sure, but in the real world, IP trafic is mostly carried over full duplex links (either serial or switched Ethernet), so you usually consider the trafic in one direction only (provided it's symetrical). If I consider an E1 to be 2 Mbps (2 inbound, 2 outbound, really), and Fast Ethernet to be 100 Mbps (again, 100+100 in full duplex), I shall consider U-law to be 83 kbits/s, not 166. -- Nicolas Bougues Axialys Interactive ___ 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] Sipura SPA 200 Fax
On Fri, Mar 05, 2004 at 03:14:58PM -0500, Mark Messmore, Technical Support, University Telcom Inc. wrote: > > Is anyone presently using the Sipura SPA 2000 for faxing? I was about > to look into it and just figured that I would ask to see if anyone ran > into any snags, problems, etc. Thanks. > No problem here (5 fax machines). Solved the ringing detection problem I had with the GS 286. -- Nicolas Bougues Axialys Interactive ___ 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] 3com NBX phones
On Thu, Mar 04, 2004 at 04:32:52PM -0500, Clif Jones wrote: > I know a little history on the 3com SIP phones... We have about a dozen > of them > where I work. I'm not familiar with the NBX100 model number but the ones we > have are labeled: P/N: 655005001. The first ones didn't support SIP out > of the > box and had to be upgraded with a new flash image. I can't recall if > they came from > the factory with H323 or the 3com proprietary IP protocol but the phones > look just > like the 3com PBX phones you see in small businesses. 3com abandoned > the phone > after spinning off the SIP division (Commworks?) and determining that > the phone > hardware just didn't have the resources to continue work on SIP. It is > a shame because > these phones boot faster than any other IP phone I have seen and have a > good speakerphone. > The image that we use is pre-RFC3261 but would probably work with Asterisk. > I have the same phones here (two of them). Got them directly from 3com in late 2000 I think. They were never distributed in the channel I think (at least not here in Europe). Mine arrived "SIP ready", and I flashed them once with a standard TFTP procedure. Unfortunatly, 3com discontinued them, no more firmware are available, AFAIK. My firmware is 1.0.1.21.0 SIP. Unfortunatly I'm not able to find this latest firmware on file. Note that the hardware is probably not the same as the standard NBX phones : my SIP phones did feature an IR sensor to be used by a Palm for automated dialing. -- Nicolas Bougues Axialys Interactive ___ 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] About Grandstream ATA-286 and ring voltage
Dear all, My GS ATA-286, which otherwise work well, seem to be unable to ring a fax (or at least, some kind of fax). The fax basically doesn't detect the ring. I measured with a volt meter about 45V during the ring pulse out of the ATA. This looks fairly low to me (supposed to be in the 70V+ range, isn't it ?). The adapter works with evey kind of phone I tried, but did not work with two different fax machines. Am I simply out of luck with these fax ? Does my ATA look defective (tried two of them, however) ? -- Nicolas Bougues Axialys Interactive ___ 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] Disabling echo cancellation on fax ?
Dear all, I'm trying to use a fax (via a SIP adapter) on an Asterisk box (going out on a Zap PRI channel). Unfortunatly, the transmission doesn't work well (if at all). Here is a "zap show channel 1" dump : Channel: 1> File Descriptor: 9 Span: 1 Extension: Context: ft_in Caller ID string: Destroy: 0 Signalling Type: PRI Signalling Owner: Zap/1-1 Real: Zap/1-1 Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: yes Pulse phone: no Echo Cancellation: 128 taps, currently ON PRI Flags: Call Actual Confinfo: Num/0, Mode/0x Actual Confmute: No The SIP side is of course using A-Law as well. The "Fax Handled" line seems to mean that somewhat Asterisk detected the fax tone. But the Echo Cancellation remains ON. Shouldn't it be disabled in such a case ? If so, did I miss a parameter somewhere ? Note that I don't have anything special in my config files for this extension : it setup just like any other local SIP phone. -- Nicolas Bougues Axialys Interactive ___ 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] Speech between Grandstream phones sounds like talking under water
On Tue, Feb 17, 2004 at 09:24:53AM -0500, Rana Dutt wrote: > I was able to solve the audio quality problem by going to > www.grandstream.com/BETATEST and downloading the latest beta firmware, > version 1.0.4.46. > This version has at least one problem I'm aware of : when you dial an external number (via an ISDN PRI), if you hangup the SIP phone the message doesn't reach Asterisk, which keeps on ringing the ISDN leg. -- Nicolas Bougues Axialys Interactive ___ 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] New Firmware for Grandstream Phones - Supports CFG by MAC address
On Mon, Feb 09, 2004 at 05:37:48PM -, David J Carter wrote: > Have a look at http://www.plugndial.com/aps_sample.html > I've been told by sipphone that this format is "new". It's not supported by anything on the market right now. -- Nicolas Bougues Axialys Interactive ___ 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] SIP phones with dual ethernet.
On Mon, Feb 09, 2004 at 11:21:42PM +0100, Tomas Prybil wrote: > > How would you "roll out" a SIP based VoIP platform to to endusers with > various connection solutions. Is there such a thing that solves the > various issues of NATting a phone? > Well, there is not "one-fit-all" solution. GS phones work well behind most NATs (even cascaded NATs). Most cheap ethernet/ethernet modems (sold as Cable/DSL routers) can do the trick. -- Nicolas Bougues Axialys Interactive ___ 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] SIP phones with dual ethernet.
On Mon, Feb 09, 2004 at 02:29:50PM -0500, Jess Magnaye wrote: > have you tried this gs-102 with pppoe? verizon dsl uses > pppoe. pppoe is No, I didn't try. Yes, pppoe is fairly standard DSL stuff (when used with an ethernet modem). > logically like dhcp, but using ppp for added feature like aaa :) can this > unit connect directly to a cable modem? > I think so. However, what is not clear at all is whether you can connect anything else to the DSL line. -- Nicolas Bougues Axialys Interactive ___ 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] SIP phones with dual ethernet.
On Mon, Feb 09, 2004 at 04:36:41PM +0100, [EMAIL PROTECTED] wrote: > > Some vendors sells phones with dual ethernet ports. Are these just > incorporating a hub/switch functionality? The reason for my question is > that the normal case for a DSL customer is the possibilty to use one MAC > adress from their service. With MAC cloning or other functionality this > could work out to be a solution. > The Grandstream 102 phone has 2 ports, with what seems to be a 10 Mbps switch inbetween. However, it does as well support PPPoE in the firmware. I'm not sure what it's meant for and/or how it actually works. I suppose the phone can sit itself right behind an Ethernet DSL modem with this setup, but I'm not sure whether/how it enables a standard PC to work on the other ethernet port. -- Nicolas Bougues Axialys Interactive ___ 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] [ot] Grandstream hardware
On Tue, Jan 20, 2004 at 08:57:32AM -0600, Eric Wieling wrote: > On Tue, 2004-01-20 at 01:12, Nicolas Bougues wrote: > > Yes, but with a Pentium you don't have to pay a license to use MMX in > your software, since the MMX instructions are part of the product you > are allowed to use them with that product. > MMX is not an algorithm. It's an instruction set. I'm not sure whether there is a patent about it, but sure Intel grants you the right to use the MMX instructions on their chips. > If I understand things correctly, the companies that make DSP chips can > implement whatever codec(s) they want and NOT have to pay the patent > holders to sell this product with the patent holder's codec in it? > DSP are really processors. Just like your favourite pentium or AMD device. The only difference is that DSP (usually) have special instructions set designed to ease the development of programs performing repetitive (often mathematical) operations on continuous data set. But for instance in GS devices, the main CPU *is* a DSP, simply because : - this DSP can do all the CPU tasks (read, non signal processing tasks) - it was the cheapest option > I ask again, how does Grandstream (from all accounts a very small > company) afford to provide the patented codecs in their products? > What's odd about that ? They are licensing software "bricks" from one or more DSP *software* vendors. These vendors probably take care of the patent issues themselves. But bear in mind that these vendors are *not* the DSP vendor (TI, in the present case). Hardware vendors sell hardware. Software vendors sell licensing, which include patent rights. That kind of agreement usually involves an upfront cost (probably in the tens of thousand dollars or so), and then a "per seat" cost. Say a few dollars per phone, for instance. No big deal. -- Nicolas Bougues Axialys Interactive ___ 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] [ot] Grandstream hardware
On Mon, Jan 19, 2004 at 05:23:26PM -0600, Eric Wieling wrote: > How does Grandstream become patent indemnified for their hardware? I > would assume they did not pay for a license for G723,1 and G729 directly > to the patent holding company. Maybe they did. I always assumed the > indemnification came with a DSP that implemented the codec. > I suppose they did pay for it. A DSP is a processor. Just like when you buy a Pentium IV, it doesn't give you the right to use, for instance, MS Windows on it. You have to pay for software. And that's what algorithms are. Except that you have to pay for algorithms even if you do your own original implementation. -- Nicolas Bougues Axialys Interactive ___ 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] [ot] Grandstream hardware
On Mon, Jan 19, 2004 at 08:44:36AM -0600, Eric Wieling wrote: > On Mon, 2004-01-19 at 02:34, Nicolas Bougues wrote: > > These are quite cheap components (the most expensive part is the $6 > > DSP). > > What *I* want to know is why someone has not made a CHEAP PCI card with > 4, 8, or 16 of these DSPs on it. This kind of card would provide > PCI boards embedding DSPs exist. However, they are not very cheap, because : - they require PCI glue (FPGA, or some sort of bridge chipset). DSPs usually can't be directly connected to the PCI bus. They probably also need some RAM, or a more complex CPU to drive them. - such existing board usually provide some kind of PCM highways, and/or switch matrices to connect to the telecom environnement - this is a small market : this drives prices up quite fast. > hardware assisted DSP functions as well as patent indemnification. > Would you even have to USE the DPSs in order to be patent indemnified? Err, I don't see the point with patent indemnification. The price you pay for the patent depends on which patents (ie, which algorithms) you use. Unless your board is limited (by firmware, for instance), to a certain set of algorithms, you can't include the price of the algorithm in the board. -- Nicolas Bougues Axialys Interactive ___ 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] configuration to Grandstream via tftp
On Mon, Jan 19, 2004 at 10:51:02AM +0100, Nicolas Bougues wrote: > > > > Attached is the config file I send to my Grandstream. > > > > Change IP address & Phone ID to suite. > > > > That's great. Is it documented somewhere ? > > And how do you manage tens or hundreds of phones ? Are they all in the > same cfg.txt file ? > Replying to myself. The GS phones use TFTP extensions (RFC 2347) to provide additional info in their TFTP requests. The server has to be aware of these extensions, if it wants to serve different files. Here is a small dump for a request from an HandyTone (key/value) : grandstream_MODEL HT-100 grandstream_NAT1 grandstream_ID 000b8200c14a grandstream_REV_BOOT 00100013 grandstream_REV_PHONE 00104026 grandstream_REV_VOC0012 grandstream_REV_HTML 00100020 grandstream_REV_VP 0010 We can easily see the MAC address, the model and the current firmware versions (1.4.26). With these informations, the TFTP server could : - serve the right cfg.txt file - serve the right firmware files (or actually, serving nothing if the server considers the phone to be up to date). I'll try to see if my basic Java knowledge enables me to make the NAT-aware TFTP server fwtftpd understand these extensions. -- Nicolas Bougues Axialys Interactive ___ 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] Connecting BRI to PRI card?
On Mon, Jan 19, 2004 at 10:44:29AM +0100, [EMAIL PROTECTED] wrote: > Hi, > > Is it possible to connect a BRI isdn line to a E100P PRI card? > > The location where I want to use it has a BRI line an will > switch to PRI in 6 month. > No, BRI and PRI are different things. The easiest and cheapest way is to buy a BRI card and then a PRI card. -- Nicolas Bougues Axialys Interactive ___ 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] configuration to Grandstream via tftp
On Mon, Jan 19, 2004 at 09:20:33AM -, David J Carter wrote: > Hans, > > Attached is the config file I send to my Grandstream. > > Change IP address & Phone ID to suite. > That's great. Is it documented somewhere ? And how do you manage tens or hundreds of phones ? Are they all in the same cfg.txt file ? -- Nicolas Bougues Axialys Interactive ___ 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] [ot] Grandstream hardware
On Mon, Jan 19, 2004 at 01:28:02AM +, Robert Murray wrote: > Hi > > Has anyone opened up a grandstream phone or handytone ATA to find out what is > inside? What is the CPU? How much RAM? > The HandyTone 286 features : - 1 Mb Flash - 256 Kb SRAM - a TI TMS320VC5402 100 MHz DSP - an RTL8019AS ISA 10 Mbit Ethernet controller - a phone interface - and some glue around that These are quite cheap components (the most expensive part is the $6 DSP). -- Nicolas Bougues Axialys Interactive ___ 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] SS7 over Asterisk ?
On Sat, Jan 17, 2004 at 04:34:34PM +0200, Alexandru Coseru wrote: > > All I'm trying right now is to get raw data from the E1 (from each > timeslot) , transmit it to another asterisk server and push it to the other > E1.. > Doesn't TDMoE do that (provided that you're on the same subnet) ? -- Nicolas Bougues Axialys Interactive ___ 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 Gear
Whoops... echo set ignore_list_reply_to = yes >>.muttrc Sorry. I believe that the Reply-To setting on this list must have been discussed here a few times here, so I won't start :) On Fri, Jan 09, 2004 at 06:00:29PM +, Adthrawn wrote: > Hi, > > I know it's not really the place, but if anybody in the UK (or US) is > interested, I'm clearing out lots of new Cisco stock... > > 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), > 7935's (conference phone) and 3550-24-PWR switches. > > I also have boxes of 7914's, the single-7914 foot stand and double-7914 > foot stand (these are required to connect a 7914 to a 7960G). > > And some useful locking and non-locking wallmount brackets for 79xx > range. > > We also have lots of PSU's for the whole 79xx range. > > I'll now feel ashamed, and sink into my seat :-) > > Best, > Ad. > > ___ > 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 -- Nicolas Bougues Axialys Interactive ___ 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 Gear
Hi, Would you mind giving me an idea of the price level for the 7970, 7960, 7940 and 7920 ? Qty 5 or more. Shipping to France. On Fri, Jan 09, 2004 at 06:00:29PM +, Adthrawn wrote: > Hi, > > I know it's not really the place, but if anybody in the UK (or US) is > interested, I'm clearing out lots of new Cisco stock... > > 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), > 7935's (conference phone) and 3550-24-PWR switches. > > I also have boxes of 7914's, the single-7914 foot stand and double-7914 > foot stand (these are required to connect a 7914 to a 7960G). > > And some useful locking and non-locking wallmount brackets for 79xx > range. > > We also have lots of PSU's for the whole 79xx range. > > I'll now feel ashamed, and sink into my seat :-) > > Best, > Ad. > > ___ > 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 -- Nicolas Bougues Axialys Interactive ___ 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] IPv6 support
On Thu, Jan 08, 2004 at 11:19:48AM +0100, Olle E. Johansson wrote: > > > >There is no real "design issue" with IPV6. No more NAT, that's all. > Oh, there is a lot of design issues if you consider a dual-IP network. > > * If you dial me from an IPv4 network and I'm on IPv6, we have to force > media proxy > (Like NAT today) > > * If you're on a IPv4 only network, dialling me and get an IPv6 address to > my SIP proxy > - what do you do? > * If you're dialing through a SIp proxy and your SDP is IPv6 and your SIP > contact is IPv4, > what do we do? > Well, usually right now : - machines have both v4 and v6 stacks - machines have both v4 and v6 addresses In this setup, there wouldn't be any change for v4 stuff. And v6 stuff is not much more complicated (although there still might be problems, like, for instance, some VoIP protocol having syntax problems with v6 addresses). The idea of a v4 client and v6 server (or proxy, or whatever) looks a little bit odd to me. If such a thing was easy to do, the world would have embraced ipv6 quite some time ago. IPv4 machines can only talk to IPv4. IPv6 maps IPv4 addresses. So a machine with an IPv6 only stack could theoritically reach a v4 client through some gateway. And I agree, this would be quite a mess from the voip protocol stand point. But hopefully, either : - v6 hosts are fully v4 enabled as well - v6 only hosts will only talk to v6 peers -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPv6 support
On Thu, Jan 08, 2004 at 08:22:26AM +0100, Olle E. Johansson wrote: > Brunner, Armin wrote: > >Is there any IPv6 support within asterisk? > >I couldn't find any hint in all the documents about this. > No. Not today. iptel.org's SER has IPv6 support. > I think it is about time to think about how to design it in Asterisk, as > many VoIP networks > based on SIP consider IPv6 to get rid of the NAT mess. > There is no real "design issue" with IPV6. No more NAT, that's all. However, network connections and DNS lookup stuff has to be IPV6 aware, as well as configuration parsing (wherever IP adresses may be expected). There might then be possible to use some IPV6 specific stuff, but it's quite far away. Note that some time in the future I could be interested in it. Our network is almost fully IPV6 enabled, and we have full, native IPV6 peering. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
OT: Re: [Asterisk-Users] Client for P800/P900
On Wed, Jan 07, 2004 at 04:28:44PM +0100, Peer Oliver schmidt wrote: > > BTW: Nicolas, are you thinking of finishing up your SyncML tool > (http://nicolas.bougues.net/syncml/) > As of now, my codebase is being used (and further developed) by various people. But I really did it as a "proof of concept", and have no more time to take care of it. Now that the wbxml library supports SyncML, it should be quite easy to make a really nice package out of it. I believe that Chuck Hagenbuch (Horde) is working on it. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Client for P800/P900
On Wed, Jan 07, 2004 at 01:42:30PM +, Andreas Anderson wrote: > Hi Guys, > > is there a client which can be used on the SonyEricsson P800/P900...? > IAX would be cool, but i take anything that can connect (via bluetooth) > to an asterisk-server ;-). > > The phone is Symbian, and can also execute java-stuff... > Would be nice. Could turn any symbian based phone into a cordless IAX phone (with limited range, though). Is the P800 able to connect to a bluetooth AP ? Or maybe you have a bluetooth suite on your PC that is able to "sense" the presence of the P800 and enable the serial-over-bluetooth link automagically ? -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Calculating Bandwith
On Wed, Jan 07, 2004 at 01:06:12AM -0800, calvis wrote: > > I am trying to calculate bandwidth needs. Is 1 T1 Line able to provide > 488.5 Gigabytes of traffic for 1 month based on a 30 day week? I did my > calculation as follows: > > 1.544 mps * number of seconds in a minute(60) * number of minutes in a hour > (60) * number of hours in a day(24) * number of days in a month(30) = > 4002048 megabits / 1024 = (3908.25 gigabits) / 8 = (488.53125 gigabytes) of > bandwidth for a T1 Line. > > Please correct me if I am wrong. > Well, if the problem is to move about 450 GB of data over your T1 on a one month timespan, it's possible. But that's probably not what you want : you probably want to establish various connections, at various times, with various remote machines. There will probably be periods during the day (at night for instance) or during the week (on week ends) when the traffic will be much less (if it's calibrated to require a T1 during the busy hours). And you don't want your line to be busy all day long : the performance will suck. The thing is, your internet link has to be able to accomodate your peak load, with measures like "I accept to have a 90% line occupation 30 minutes per day" in order to have a fairly decent quality. About 200 GB a month (total of both directions) would qualify, IMHO, as "optimised bandwidth usage" in that kind of context. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
On Sun, Jan 04, 2004 at 07:38:16PM +, WipeOut wrote: > > Also a failover system would typically only be 2 servers, if there were > a cluster system there could be 10 servers in which case five 9's should > be easy.. > Err, no. five 9s is *never* easy. Does your telco provide you with SLAs that make five 9s reasonable at all ? Do you really need five 9s ? There is no such thing I'm aware of in enterprise grade telephony. You have to go to "carrier grade" equipment, which asterisk, and PCs in general, are definetly not aimed at. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] License questioni supose ??
On Fri, Jan 02, 2004 at 12:23:52PM +0100, Michael Devenijn wrote: > I have some strange question bout the asterisk (gpl license ...) but i'm not an > experienced linux user ... > > What happens if for example a big company buys digium , do we have a garantuee that > asterisk stays opensource ??? > Until the last released version, yes. Digium owns the Copyright : they can decide whenever they want that their next release will have any other kind of licence (open or not). But what's released under the GPL stays so. Anyone can continue using it, or even re-release it, provided that they still comply to the GPL terms. Nobody, even Digium, can revoke the GPL licence for GPL-ed software. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Current database abstraction effort ?
Dear all, I read across Asterisk's lists archives, and found out various discussions about how nice it would be to have a (SQL) database abstraction layer enabling the use of various SQL backends, for various purposes inside of Asterisk. As far as I see, there is no such thing yet, although there are various efforts (on CDR and voicemail, mostly) to use external databases. As I'm planning some work that will require external database support (in order to be fully dynamic and potentially shared by different Asterisk servers), I have to solve this issue first. It's definetly not a major issue, but before starting to work on it I'd like to know what the community has to say about it. I found a few references to the way the FreeRadius people did it, and I would probably make something similar, although there a a few things in their design that I would make otherwise. Finally, Asterisk uses the db1 library to store some local dynamic data. The db.h interface could be kept as-is and optionnally work with this new database backend for tasks that do not require full SQL syntax (although a there shall be a way to specify if the caller is expecting its data to be local or shared amongst servers). -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Backup Proxy & Automatic Failover
On Tue, Dec 30, 2003 at 06:49:51PM +, Adthrawn wrote: > Hi, > > The term TDM is banded around too, but from my knowledge, TDM is > trunking (probably some clever acronym relating to trunking), and in > Asterisk's case, using the IAX protocol. This leads me to the big > question; > TDM is time division multiplex. It's how phone calls are sliced on digital lines. FDM is frequency domain multiplex : to have two (or more) phone calls on the same wire, you shift the frequencies used for each call. That's similar to the way ADSL can work at the same time as telephone. TDM is far more efficient : you digitalize the phone data (sound), it gets you 8000 bytes per second (64kbps/s). Then, you take a T1 link for instance, you put on it a carrier that allows you to transfer 1536 kbits/s. And you realize that every 8000th of a second, your T1 can transport 24 bytes. This way, every 8000th of a second, you send 24 bytes from 24 simultaneous phone calls. That's TDM. Asterisk (zaptel) does TDM over Ethernet as well : instead of using a sync link such as a T1 or E1, you send multiplexed frames across the Ethernet. > Is there anyway of shifting the load of one Asterisk server to another > without breaking or loosing a call? > No. If you're talking about ISDN, that would mean asking the telco to transfer the call from one span to another, transparently. If you're talking about VoIP, that would mean doing tricky re-routing, which could be nasty if NAT is involved. In any case, that would mean transfering a whole call context as well, like a running AGI script for instance, which is simply not possible. > I know that with Survivable Routing (Cisco's big on this), the ISDN > interface is actually a router; so the Proxy is just used to decide the > destination and LCR functions, and then hands off to a router. This of > course, if a Proxy went down, would just prevent new calls from being > made, whilst existing calls can continue merrily - until someone > switches the Router off, or corrupts the IOS settings :-) > What if the "ISDN router" goes down ?? Asterisk is much more of an ISDN router kind than a proxy kind. Maybe you could try to use SER as a front end ? -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk servers sharing/propagating registry ?
On Tue, Dec 09, 2003 at 08:40:36AM -0800, Chris Albertson wrote: > > It would be a major change to the code but I think what you'd > want to do is have the Asterisk server store _all_ of it's information > in something like a database, The dail plan, SIP registrations, > everything would have to go there. Once you've done that any > number of Asterisk servers could share the same database and there > are methods of running mirrored databases already. > This is one way of doing it. The other way being to exchange live routing/registration data, and keep and process it locally, somewhat like what a BGP router does. > When I worked at a dot.com we had a design requirement that I > should be able to go into the server room and pull any of the AC > power cords and the users should not be able to know. About the > _only_ way to do this is with "load balancing". Fail over does > not work so transparently. > It's not the goal here. I'm not looking for "five nines". I'm looking for a way to : - connect several tens of E1 channels (thus I need a quite a few different boxes) - support the fact that any single box may be down. I could drop the channels it currently processes, which is OK. But when the user dials again, it should work. It should be fairly OK with the client performing a new DNS lookup before it registers, if the DNS is aware of which box is up. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk servers sharing/propagating registry ?
On Tue, Dec 09, 2003 at 08:02:18AM -0600, Mark Spencer wrote: > I suppose trunk groups on SIP would be interesting. > As I understand, trunking in IAX is meant to transport voice packets from several calls between two hosts in the same lower layer packet. Are there registration features related to trunking ? In fact, what would be very nice to have would be some kind of trunking with registry/dialplans automatic exchange, so that one can easily setup a larger virtual PBX, that would server both capacity and redundancy requirements. I keep on thinking :) -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk servers sharing/propagating registry ?
On Tue, Dec 09, 2003 at 08:28:27AM +0100, Florian Overkamp wrote: > > Registration cascading is not possible (I think) but could it be solved > with a shared dial route: > > Instead of DIAL(IAX/sip.isp.com) could you not > DIAL(IAX/sip1.isp.com&IAX/sip2.isp.com&IAX/sip3.isp.com) to reach a similar > effect ? (or chain them in different lines so it tries to reach the first > one, then the second one if it fails, and the third if that fails. > > Florian > This could solve the problem in the short term, but is not really "elegant". Each time I add a server I'd have to modify quite a lot of extensions.conf on each server. I'm not quite familiar with IAX(2) and registration questions. Does anybody sees any tricky problem that could arise with some kind of auto-cascading registrations through IAX (that is, any registration, SIP, IAX... would be forwarded on an IAX channel) ? -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Asterisk servers sharing/propagating registry ?
Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com (thus ending on a random asterisk servers) - but after that, all the servers would be "aware" of the registration. Thus any asterisk server would know how to route a call to SIP/ Same thing for IAX peers. Of course, setting up various IAX links between each server is no problem (with registration cascading, for instance). Is such a setup common, if at all possible ? -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP (peer to peer?)
On Mon, Dec 08, 2003 at 10:20:43PM +0100, Brancaleoni Matteo wrote: > SIP control messages goes always through the server > (port 5060) , only RTP media streams is p2p . > > you can see RTP passing not p2p but by * server if: > * the phone doesn't supports reinvites > or > * set in sip.conf canreinvite=no in the user definition > Or of course, if Asterisk thinks that it needs to process the stream : for instance, if you want Asterisk to be able to transfer your call (t/T options for Dial). -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prefix the * character
On Mon, Dec 08, 2003 at 08:58:07AM -0600, Barton Hodges wrote:
>
> The lines in a context get reordered. If you want to force the order
> of those lines, put the exten lines in separate contexts and "include"
> them... something like this:
>
> [some-context]
> include => prefix
> include => my_local_e164_extension
>
> [prefix]
> exten => _,1,Prefix(3312345)
>
> I don't know if that will solve your problem, but it is something to
> consider.
>
My problem is that the exten lines in "my_local_e164_extension" still
have to start at 2, since "prefix" used the 1 position, and that's
what I'd like to avoid by using "s".
To do that, I put "immediate=yes" on my PRI in zapata.conf, but
unfortunatly the Prefix command will use "s" as the extension, and
generate a new extension like 3316918s, which is not really nice.
Is there any way ta manipulate ${EXTEN} as a variable, rather that
wich the Prefix function ? If so, I haven't found out.
--
Nicolas Bougues
Axialys Interactive
___
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Re: [Asterisk-Users] Prefix the * character
On Sun, Dec 07, 2003 at 10:57:33PM -0600, Tilghman Lesher wrote:
>
> Prefix is an older application which was more useful prior to being
> able to manipulate variables (the days of BYEXTENSION instead of
> ${EXTEN}). Instead, do:
>
> Dial(SIP/[EMAIL PROTECTED])
>
I have a smilar problem : I have a default context for an interface,
where I'd like to prefix all incoming calls DID numbers (basically,
the telco sends the last 4 digits dialed, I want to fully qualify my
E164 number before doing extensions processing).
I don't know much (yet!) about Asterisk, so I thought something like
exten => s,1,Prefix(3312345)
include => my_local_e164_extensions
would do the trick. Unfortunatly, if the ${EXTEN} was 6060 at that
time, I get a new extension as s6060 (instead of 33123456060). Is it
supposed to be this way ?
So instead I had to do something like
exten => _,1,Prefix(3312345)
include => my_local_e164_extension
which works fine, except that now I'm at the "2" level in the context,
and I had to modify my_local_e164 extension context accordingly.
Does somebody know of a better way to do it ?
Thanks.
--
Nicolas Bougues
Axialys Interactive
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Re: [Asterisk-Users] GrandStream Budgetone Phone & DHCP & General Observations
On Sat, Dec 06, 2003 at 04:35:55PM -, David J Carter wrote: > Hi Nicolas, > > Thanks for the file. > > I would appear to have some of the file missing that the BT-100 is looking > for. > > Ala, cfg.txt > sipp.bin > ring.bin > > After the tftp update the program is still showing 1.0.3.81. > > Any thoughts. > > The problem is, as I stated in an earlier email, that the TFTP client in the GS phone is somewhat odd, and will only accept files from "NAT enabled" TFTP server. As far as I can see, the TFTP server shall send the data from its port 69, which is not what the normal ones do. I'm not sure whether this is RFC compliant. So yes, I have the 1.0.4.17 FW files, but I have no way to have the phone download them ! I'll try to hack a "NAT friendly" tftp server on monday. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream Budgetone Phone & DHCP & General Observations
On Fri, Dec 05, 2003 at 04:40:42PM +, Michael T Farnworth wrote: > On Fri, 5 Dec 2003, Nicolas Bougues wrote: > > > On a slightly different topic : does somebody know of a NAT-friendly > > (as Grandstream means it) tftpd server ? It seems theirs replies from > > port 69, which is the only thing their phones will accept. > > > > [ If anybody wants it, I can send the 1.0.4.17 firmware by email ]. > > I am slightly puzzled why the 1.0.4.17 firmware isn't the version that > Grandstream offers through their tftp if it is the latest version. I just > noticed that my new Grandstream phones have 1.0.4.17 whilst the older ones > have 1.0.3.81. > > Can anybody shed any light on the reason? > I've been told by TelAppliant it would soon appear on their website / TFTP server. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream Budgetone Phone & DHCP & General Observations
On Fri, Dec 05, 2003 at 10:42:02AM -0500, Glenn Dalgliesh wrote: > Symptom: Phone after about 15mins will stop functioning > Problem: DHCP lease renewed but default route dropped > Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is > released > > It turn's out that these phones have a few issue in 1.0.3.81 > firmware. The phone may stop transmitting packets if configured with > DHCP, if DHCP is being provided by certain devices. Netopia routers > have been confirmed in this category. It turns out that there is > some differences btw the implementation of DHCP btw different vendor > and this is causing the phone to loose it default route and stop > transmitting packets approx 15mins after the phone receives it's > lease after reboot. GrandStream says this will be fixed in the next > release. > Interesting. We have 6 GS phones, one is 1.0.3.81 and has this behaviour, the others, 1.0.4.17 are ok. The DHCP server is Linux dhcpd. In a remote office, they have an Allied Telesyn router providing DHCP, and all the phones, no matter the version, work well. On a slightly different topic : does somebody know of a NAT-friendly (as Grandstream means it) tftpd server ? It seems theirs replies from port 69, which is the only thing their phones will accept. [ If anybody wants it, I can send the 1.0.4.17 firmware by email ]. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Port density: DS3 cards?
On Thu, Dec 04, 2003 at 02:53:43PM -0600, Steven Critchfield wrote: > On Thu, 2003-12-04 at 14:06, John Todd wrote: > > Obviously, there are no DS3 TDM cards that are currently compatible > > with Zap channels. (or are there?) > > > > Does anyone know of an inexpensive DS3 card that could perhaps be > > used with Asterisk if one were to try to port the Zap drivers to such > > a card? PCI, of course, would be the bus of choice. > > Your first problem will be bus speeds. A single DS3 is 44.736Mbps. each > way. So if you double this and get the 89.472Mbps, you are going to be > coming close to the real limits of the 32/33mhz PCI bus without having > done any work on the data you are shuffling. So while it could be done > here, I'd start worrying about stability. Sure you could switch up to > faster PCI buses like the 64/66mhz bus, but then you will start limiting > what systems you can use. Then again, if you are putting that many > channels through a single machine, there wasn't many choices for the > hardware to begin with. > Don't get confused by Megabits and megabytes. A 32/33 PCI bus has a max (theoritical) bandwidth of 133 Megabytes per second. That's about 1 Gigabit. So as long as you don't do much other things on the bus (for latency concerns), 89 Mbps is ok on standard PCI. Channelized DS3 boards do exist, and writing a zaptel driver for it should not be too hard. What I'm not sure, though, is whether telcos provide it to end users for voice services. At least here in Europe, the telco would rather run the DS3 to your building, then split it up in a bunch of E1 on their own CPE. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS over PRI/E1?
On Wed, Dec 03, 2003 at 08:30:34AM +0100, Roy Sigurd Karlsbakk wrote: > hi all > > I spoke to this guy the other day, working with Cisco's VoIP system. He > told me they were using a PRI/E1 to transport SMS, and could even do so > from their phones. > > May this be possible with asterisk? I have an E100P in my primary > asterisk server connected to a E1/PRI. > This is a carrier service. I seem to remember there are some ETSI standards for SMS over ISDN transport/gatewaying. As far as I understand (which is not very far, I admit, right now!), Asterisk doesn't support Q931 user-to-user info transmission (although there is support in libpri). It should be possible, however, to modify chan_zap and add a SendText application (and/or modify the Dial app) to handle this. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Core voice prompts in french ?
Dear all, I'd like to know if the "core" (demo, voicemail...) asterisk prompts have ever been recorded in french (and are freely available). If not, I'm willing to have them studio-recorded by a professional speaker, and contributed back to the community. Does a message list exist ? -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Party in Paris
On Sun, Nov 30, 2003 at 01:36:09PM -0600, Mark Spencer wrote: > I'll be there until jan 5. The 19th would definitely be too early, maybe > the 20-22? Possibly even after the new year, jan 2 or 3. > > Mark > That's great news, I was sorry I would be abroad on the 19th. I would definetly be ok for a meeting in early january. If needed I can arrange just about anything, a drink, a dinner, and/or hosting a meeting at my company. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancellation
On Wed, Nov 19, 2003 at 06:56:04PM -0600, Rich Adamson wrote: > > I've got an X100P & a cisco 7960. if i call from an analog line via the > > x100p to the cisco, there is an overly audible echo on the cisco. If i > > make a call from a cisco to cisco, there is no echo. zapata.conf has > > echocancel=yes & echocancelwhenbridged=yes set. Any ideas? > > > > I'm currently using the default implementation of echo > > cancellation...which one should I try next? > > > > Take a look at: > http://www.voip-info.org/wiki-Asterisk > > Most echo problems are not actually related to the X100P or the software, > but rather the pstn line (including house wiring, analog phones, and other > crap left hanging on the pstn line). It's a technical issue that can't be > resolved with non-technical approaches. > It's called line echo. Basically, the hybrid that converts the 4 wires (2 TX/ 2 RX) into a 2-wire analog phone line intrinsically generates echo. It's not noticeable (you hear your own voice while you talk, just like your ear hears your mouth in a plain talk) when the switching is done at the byte level in the TDM mux. But it does exist. That's why faxes are half duplex devices, and full duplex modems incorporate full duplex echo cancellers. Digital cell phones, digital phones (either ISDN or connected to proprietary PBXs) or IP phones don't generate line echo. And they are usually clever enough to avoid acoustic echo as well. But then, in the case of line echo, removing it is not an easy task. It involves quite complex signal processing. Look at the various tries in the zaptel source. An no, POTS phones don't have echo cancellers. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
