[Asterisk-Users] BUG: chan_oss outgoing calls autoanswer=no

[Klaus-Peter Junghanns]

Hi,

today i setup my desktop box as an asterisk server
to use chan_oss for my headset (gnophone and my soundsystem
give me huge delays). in oss.conf i set autoanswer=no
because i dont want incoming callers to eavesdrop on me
when i am singing loud to my ABBA mp3s ;-)

incoming calls work fine, i type "answer" on the console
and the call is connected. but outgoing calls are somewhat
odd:
- i do a "dial 17001578573"
- oss answers the channel i hear the "pling"
- app_dial starts
- and continues ringing indication forever
  although the other party can hear me already
- when i type "answer" (remember it's an outgoing call)
  i get connected and can hear them, too.

Mark, is this the wanted behaviour of chan_oss? or a small bug?

regards
kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
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Re: [Asterisk-Users] DTMF with IConnectHere fix

[Klaus-Peter Junghanns]

Hi there,

to stop my inbox from being filled with those "...iconnect
inband dtmf .. yadda yadda..." mails, i am sending this patch 
to the list ;-)
thanks to lele for updating it to patch against latest cvs :)
you can enable inband dtmf by adding the line inbanddtmf=1
to the sip pvt in sip.conf.

Mark, i will send you a disclaimer on monday (really!), hope
you will merge it (although you dont like the idea...).

regards
kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]

Am Fre, 2003-02-28 um 18.08 schrieb Mark Spencer:
> I am looking for someone to confirm that Iconnecthere sends DTMF either
> in-band or out-of-band.  If it is out-of-band, then I'll need to work with
> someone to figure out the issue.  If it is in-band then I will need to add
> in-band DTMF detection to SIP.
> 
> Mark
> 
> On Thu, 27 Feb 2003, Ben Clark wrote:
> 
> > Is anyone working on a fix to send / receive DTMF with IConnectHere?
> >
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--- asterisk.wml-T50/channels/chan_sip.c2003-02-27 09:10:52.0 +0100
+++ asterisk.wml-T50-dtmf/channels/chan_sip.c   2003-02-28 01:35:35.0 +0100
@@ -33,6 +33,7 @@
 #include 
 #include 
 #include 
+#include 
 #include 
 #include 
 #include 
@@ -170,6 +171,9 @@

int maxtime;/* Max time for first 
response */
int initid; /* 
Auto-congest ID if appropriate */
+
+int inbanddtmf;
+struct ast_dsp *vad;

struct sip_peer *peerpoke;  /* If this calls is to poke a 
peer, which one */
struct sip_registry *registry;  /* If this is a REGISTER call, 
to which registry */
@@ -197,6 +201,7 @@
int amaflags;
int insecure;
int canreinvite;
+int inbanddtmf;
struct ast_ha *ha;
struct sip_user *next;
 };
@@ -216,6 +221,7 @@
int capability;
int insecure;
int canreinvite;
+int inbanddtmf;
struct sockaddr_in addr;
struct in_addr mask;
 
@@ -368,6 +374,7 @@
r->insecure = p->insecure;
r->canreinvite = p->canreinvite;
r->maxtime = p->maxms;
+   r->inbanddtmf = p->inbanddtmf;
strncpy(r->context, p->context,sizeof(r->context)-1);
if ((p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) &&
(!p->maxms || ((p->lastms > 0)  && (p->lastms <= 
p->maxms {
@@ -671,6 +678,9 @@
needcancel = 1;
/* Disconnect */
p = ast->pvt->pvt;
+if ((p->inbanddtmf == 1) && (p->vad != NULL)) {
+ast_dsp_free(p->vad);
+}
p->owner = NULL;
ast->pvt->pvt = NULL;
 
@@ -916,6 +926,10 @@
else
snprintf(tmp->name, sizeof(tmp->name), "SIP/%s:%d", 
inet_ntoa(i->sa.sin_addr), ntohs(i->sa.sin_port));
tmp->type = type;
+if (i->inbanddtmf == 1) {
+i->vad = ast_dsp_new();
+ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT);
+}
tmp->fds[0] = ast_rtp_fd(i->rtp);
ast_setstate(tmp, state);
if (state == AST_STATE_RING)
@@ -1035,6 +1049,9 @@
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
+if (p->inbanddtmf == 1) {
+f = ast_dsp_process(p->owner,p->vad,f,0);
+}
}
}
return f;
@@ -2458,6 +2475,7 @@
strncpy(p->accountcode, user->accountcode, 
sizeof(p->accountcode)  -1);
p->canreinvite = user->canreinvite;
p->amaflags = user->amaflags;
+p->inbanddtmf = user->inbanddtmf;
}
break;
}
@@ -2489,13 +2507,18 @@
if (p->owner) {

Re: [Asterisk-Users] DTMF with IConnectHere fix

[Klaus-Peter Junghanns]

hi there,

mark said i need a wishlist and a personal page, see?:

[21:35:44]  kram thinks kape should have a wishlist too!
[21:35:59]  kapejod already has one, but nobody ever found it ;)
[21:36:11]  ah
[21:36:27]  maybe i need a personal web page, hehe ;)
[21:36:33]  *nods*

his wish is my command ;-) ... so here it is:

http://62.8.140.130/~kapejod/kapejod.html
[ just to have it in the archives ;-) ]

regards
kapejod

Am Sam, 2003-03-01 um 21.27 schrieb Mark Spencer:
> > Mark, i will send you a disclaimer on monday (really!), hope
> > you will merge it (although you dont like the idea...).
> 
> You're right, I don't like the idea of *using* in-band DTMF on SIP, but I
> do recognize the necessity in supporting it.
> 
> Your patch is very well done and I'll be happy to merge it.  I will
> contact you off-list about some related ideas.  Thanks!
> 
> Mark
> 
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Re: [Asterisk-Users] Using asterisk with ISDN - How to configure

[Klaus-Peter Junghanns]

Hi Cristoph,

the B1 has linux capi 2.0 drivers, so you can use chan_capi.
Grab it at http://www.junghanns.net/chan_capi.html
read INSTALL and README, take a look at the sample configuration
file capi.conf.

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET Internet-Services & Software-Development GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
mobile: +49 160 7503372
email:  [EMAIL PROTECTED]


Am Mon, 2003-03-03 um 21.56 schrieb Christoph Schütz:
> Hello,
> 
> first of all I have to say: I read the handbook but there are some questions 
> left I couldn't answer. It's very basicaly. Sorry about that.
> 
> I have a AVM B1 PCI - Card. I think there is a I4L and a Capi-Port installed.
> 
> 1.  How can I define the right channels, how are they defined. I found the 
> modem.conf example with an ISDN section but how can I make asterisk load it?
> 
> 2. Do I have to compile Zaptel, zapata or libpri ?
> 
> 3. How can I dial, if I have configured anything. Is it only possible by using 
> the console or is there any frontend?
> 
> If there is somebody out there who solved this and has configuration files, I 
> would be pleased to get them as an example. 
> 
> Thanks for help Christoph
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Re: [Asterisk-Users] g.711 to GSM gateway

[Klaus-Peter Junghanns]

> Hello:
> 
> I would like to implement a VoIP telephony system; client's
> software/hardware only supports SIP with the g.711 codec.
> 
> It's very huge in wan communications (about 110 Kbps) !!!
> 
> I'm looking for systems that could reduce this bandwidth, at this time I
> can't change the client side.
> 
> Can Asterisk act as a gateway to convert g.711 to GSM, and then reduce the
> wan traffic?. The idea is on-the-fly conversion:
> 
> client 1 -> Asterisk GW 1 -> wan -> Asterisk GW 2 -> client 2
> 
> client 1 calls client 2 and sends g.711
> Asterisk GW 1 converts on-the-fly g.711 to GSM
> Asterisk GW 2 converts on-the-fly GSM to g.711
> client 2 receives g.711 from client 1
> (and vice-versa)
> 
> Is it possible to do it with Asterisk?
> 

YES!

> Thanks in advance.
> 
> Best regards,
> Cerrajetto.
> 
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regards
kapejod

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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
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[Asterisk-Users] CeBIT

[Klaus-Peter Junghanns]

Hi there,

just wanted to let you know that i will be wandering around on
CeBIT this Friday the 14th. If anybody wants to meet and bug me with
some capi, isdn or general asterisk stuff contact me off list.
If not i will sit silently in the corner and wait for my train
back to Berlin  ;-)

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
mobile: +49 160 7503372
email:  [EMAIL PROTECTED]


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Re: [Asterisk-Users] Can an X100P be used as an FXS?

[Klaus-Peter Junghanns]

> Hi All...
Hi Jim
> 
> Can an X100P be used as an FXS?
> 
no.
> Thanks...
> 

 regards
kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


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[Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

hi alaw (and now also ulaw) folks,

version 0.1.0 is out now. thanks to Bicster for beating
eicon capi to support ulaw and echo cancellation!!

http://www.junghanns.net/chan_capi.html

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

Hi Liaan,

looks like you dont have capi4linux installed. What card
are you trying to use?

regards
kapejod


-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]



Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> Hallo
> Thanks very much for great work
> Just quick Q.
> Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> Line 1984 (load_module): CAPI not installed!" when starting???
> Thanks
> 
> 

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Re: [Asterisk-Users] CAPI errors...

[Klaus-Peter Junghanns]

Morning Roy,

capi says that the send queue is full. what did you stick in there? ;-)

regards
kapejod

Am Don, 2003-03-13 um 12.35 schrieb Roy Sigurd Karlsbakk:
> hi
> 
> I keep getting these errors all the time:
> 
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[3076]: File chan_capi.c, Line 1081 (pipe_msg): error sending 
> DATA_B3_RESP (error=0x1102)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> ERROR[74780]: File chan_capi.c, Line 735 (capi_write): error sending 
> DATA_B3_REQ (error=0x1102, datalen=160)
> 
> anyone that might know why? Klaus?
> 
> roy
> -- 
> Roy Sigurd Karlsbakk, Datavaktmester
> ProntoTV AS - http://www.pronto.tv/
> Tel: +47 9801 3356
> 
> Computers are like air conditioners.
> They stop working when you open Windows.
> 
> 
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

Hi,

this is funny :)
you expect it to work with no card 
i dont think that there are external TAs with capi 2.0
linux drivers. isdn4linux wont help much...you want
capi4linux, but without a card it's not much fun ;-)

regards
kapejod


Am Don, 2003-03-13 um 11.11 schrieb Liaan van der Merwe:
> Hallo
> To be honest.. no card as yet.
> Truying to find out whether external TA will work.
> I'll install latest i4l and see what happens
> Thanks
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter
> Junghanns
> Sent: 13 Maart 2003 11:54
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
> 
> Hi Liaan,
> 
> looks like you dont have capi4linux installed. What card
> are you trying to use?
> 
> regards
> kapejod
> 
> 
> --
> Klaus-Peter Junghanns
> 
> CEO,CTO
> Junghanns.NET GmbH
> Breite Strasse 13 - 12167 Berlin - Germany
> fon:+49 30 79705390
> fax:+49 30 79705391
> iaxtel: 1-700-157-8753
> email:  [EMAIL PROTECTED]
> 
> 
> 
> Am Don, 2003-03-13 um 10.23 schrieb Liaan van der Merwe:
> > Hallo
> > Thanks very much for great work
> > Just quick Q.
> > Why would asterisk keep on telling me :" NOTICE[8192]: File chan_capi.c,
> > Line 1984 (load_module): CAPI not installed!" when starting???
> > Thanks
> >
> >
> 
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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

> Liaan van der Merwe <[EMAIL PROTECTED]> schrieb: 
> >Oops... see. I need to learn bit more about this whole capi thing.
> >It will be a planet external isdn TA.. capi2.0 compatible.
> 
> But not under Linux. The only card that support capi under linux are cards
> from AVM and Eicon.
> 
> It want work with external TA. If you need a external box, there will be USB
> ISDN card from AVM available, which will be supported by capi.

somebody tried that usb AVM thing, without much success,
please forget this usb thing ;-)

regards
kapejod

> 
> Cu
> --
> Lars Kneschke
> http://www.kneschke.de
> written with FeLaMiMail
> http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif
> 
> 


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RE: [Asterisk-Users] chan_capi version 0.1.0 released

[Klaus-Peter Junghanns]

CAPI == Common isdn API

it's not about talking to certain modems, it's a hardware
independent interface to access all kinds of isdn controllers
in the same way.

regards
kapejod

Am Don, 2003-03-13 um 15.28 schrieb Liaan van der Merwe:
> Ok... now I'm lost
> What is the idea then behind capi2.0 standard??
> Is this not a "language" to talk to certain modems?
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Lars
> Kneschke(priv.)
> Sent: 13 Maart 2003 15:39
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] chan_capi version 0.1.0 released
> 
> Liaan van der Merwe <[EMAIL PROTECTED]> schrieb:
> >Oops... see. I need to learn bit more about this whole capi thing.
> >It will be a planet external isdn TA.. capi2.0 compatible.
> 
> But not under Linux. The only card that support capi under linux are cards
> from AVM and Eicon.
> 
> It want work with external TA. If you need a external box, there will be USB
> ISDN card from AVM available, which will be supported by capi.
> 
> Cu
> --
> Lars Kneschke
> http://www.kneschke.de
> written with FeLaMiMail
> http://www.saunalahti.fi/sakarit/kerro-lisaa/bart.gif
> 
> 
> 
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Re: [Asterisk-Users] chan_capi: advice needed on isdn card

[Klaus-Peter Junghanns]

Morning Chris,

1. it's not predictable if you will get echo with the passive AVMs
(and probably also not predictable with the actives).

2. yes.

3. the passive Eicons (those without the word "server" in the name)
have no echo cancelation and not even a capi driver (so you're stuck
with i4l). but not all active Eicons support echo cancelation!
Bicster: can you shed some light on this? :-)

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


Am Fre, 2003-03-14 um 22.20 schrieb Chris Wetemans:
> Since chan_capi now supports echo_cancellation on (some) eicon cards, i'm
> considering buying another isdn-card.
> 
> On the moment I'm using an old Teles card with isdn4linux, but i get a
> terrible echo when calling analog counterparts, and the delay is also quite
> heafty.
> 
> 1. If I get a (cheap) AVM-card (Fritz), and use CAPI, would the delay
> (latency) be so small that an echo isn't noticeable anymore?
> 
> 2. If the echo would still be noticeable would an EICON-card with echo
> cancellation on board help a lot?
> 
> 3. Which EICON-cards have echo cancellation and linux CAPI-support, the
> cheaper client cards( DIVA Pro, DIVA+CT,...) or the expensive Server cards
> (BRI-server, BRI-server voice, )
> 
> 
> Thanks, Chris
> 

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Re: [Asterisk-Users] Voicetronix

[Klaus-Peter Junghanns]

hi brian,

the wildcard s400p from digium will be available with 4 FXS ports.

regards
kapejod

Am Don, 2003-03-20 um 22.19 schrieb Brian J. Schrock:
> Has anyone gotten the voicetronix boards to work with Asterisk, what 
> would it take? Or does anyone know where I can get 4 ports or more fxs 
> PCI cards that do work with asterisk?
> 
> Brian J. Schrock
> Network Engineer, RHCE, CCNA
> Anistone Technologies
> Phone: 614-798-9106
> FAX: 614-573-7165
> 6926 Avery Rd.
> Dublin, OH 43017
> 
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Re: [Asterisk-Users] T4000P vs. Tormenta 2

[Klaus-Peter Junghanns]

nopers, the torisa is the isa card.
the tormenta 2 is the t/e400p, if you want to do some
home soldering you can get the specs form zapatatelephony.org.
since january the tor2 specs are online.


Am Mon, 2003-03-24 um 21.04 schrieb Steven Critchfield:
> On Mon, 2003-03-24 at 11:40, d hinton wrote:
> > hi there's a few things i REALLY need to know
> > the T4000P is based on the Tormenta 2 so what is the differance?
> 
> The T400P is PCI and the tormenta 2 is ISA. The tormenta2 is a home
> build card, the T400P is factory made. T400P comes with an hour of
> support by Digium. 
> 
> > a few things more;
> > 1. is there a differance between the two?
> > 2. if so what are the the differances?
> > 3. which one is more stable? based on changes and testing :-)
> 
> The PCI T400P is less susceptable to chipset speeds as there is way more
> than enough overhead for the data on the PCI bus.
> 
> > 4. is the Tormenta 2 FCC approved. 
> > did a search on their (FCC's) db, and found nothing for Tormenta 2 or
> > the T4000P. this is not good because the cards can't be legally
> > connected to the PSTN without FCC cert. and my insurance will not
> > cover us without certification.
> 
> http://www.marko.net/asterisk/archives/0204/0134.html
> T400P approved, but no numbers mentioned.
> 
> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
> 
> ___________
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] CE certification for Europe

[Klaus-Peter Junghanns]

Hi Vlasis,

CE is no certification, it is just a decleration of conformity
from the manufacturer. It has nothing to do with getting an
ITU / ETSI (whatever...) approval for communication equipment.

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


Am Die, 2003-04-01 um 10.13 schrieb Vlasis Hatzistavrou:
> Hello,
> 
> I'd like to ask if there are any news about CE certification of the E1
> boards. I know that the T1 boards are FCC certified but I'd also like to
> know what is the status for CE certification.
> 
> Thanks for any input,
> Vlasis Hatzistavrou.
> 
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Re: [Asterisk-Users] * on openmosix

[Klaus-Peter Junghanns]

Morning Liaan,

i thought about it a while ago, but i wont work.
openMosix can distribute process over the cluster, but
for this to work you need a real multi-process application.
since * is a multithreaded application all * processes
will stay on a single node.

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


Am Mit, 2003-04-02 um 08.44 schrieb Liaan van der Merwe:
> Hallo all
> Just for interest sakes..
> Have any of you tried to run * on a openmosix cluster?? Could be nice for
> ppl like me that don’t have the resources to buy “proper” machines (I can
> get P1 166 for almost nothing, but higher than that is to expensive)
> Thanks
> Regards,
> Liaan W van der Merwe
> 
> GGATES NETWORK SOLUTIONS
> We do not only follow best practices, we invent them!
> m: +27 (0)82 9404 128
> t: +27 (0)21 8513549
> u: www.ggates.co.za
> e: [EMAIL PROTECTED]
> 


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Re: [Asterisk-Users] CE certification for Europe

[Klaus-Peter Junghanns]

Hi d hintion,

hmmm...getting approvals for europe isnt that easy.
because you get the approval for a combination of hardware
and driver software, so when you change the driver you loose
the approval.

oh yes, sure you can produce the cards and sell them cheaper,
but that doesnt take the development time of the zaptel drivers
into account. opening up a competition against digium based on
their software and GPLed hardware design doesnt sound good to me .
rather sounds like M$ style to me.

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]


Am Don, 2003-04-03 um 17.15 schrieb d hinton:
> you know, i asked diguim guys for this info and got nothing. so my company
> decided to produce these cards for our own use and have them tested for FCC,
> CA and CE certs
> 
> est. for testing we got back, range between $7,500 - $9,500 (USD) for all
> three.
> we are also surprised that the quote we got for producing these cards was so
> cheap, that we could produce them and sell them for just under $850(USD). we
> believe that this would be more in line with the reason zap tel guy's
> released the plans GPL. so that average developers could afford the card. if
> there's enough intrest i'm sure i could get my boss to sponsor a project
> that provides driver support for wider use of the zapata card and lower cost
> hardware, unless diguim wishes to do it (HINT). shout out to me before the
> end of this week, cause our cards go to the manufacturer on 4/10/2003 and
> the testing starts no soon as the manufacturer sends us the prototype. the
> total led time we got back was 3-6 weeks.
> dwayne
> - Original Message -
> From: "Michiel Betel" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Cc: <[EMAIL PROTECTED]>
> Sent: Thursday, April 03, 2003 8:18 AM
> Subject: RE: [Asterisk-Users] CE certification for Europe
> 
> 
> > I called around and got some rough quotes for R&TTE testing and
> > certification for europe. It seems to boil down to euro 2400,- per card to
> > be tested. They would also need the tech. doc and design from digium...
> >
> > Any european users want to help? I'd like to be able to legally use the E1
> > cards
> >
> > Michiel
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Vlasis
> > Hatzistavrou
> > Sent: woensdag 2 april 2003 15:25
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] CE certification for Europe
> >
> >
> > Thank you very much for your reply and for clarifying this point.
> >
> > Does anyone know if there is any effort for approval of the boards as
> > communications equipment in Europe then?
> >
> > Best regards,
> > Vlasis.
> >
> > Klaus-Peter Junghanns wrote:
> >
> > > Hi Vlasis,
> > >
> > > CE is no certification, it is just a decleration of conformity from
> > > the manufacturer. It has nothing to do with getting an ITU / ETSI
> > > (whatever...) approval for communication equipment.
> > >
> > > regards
> > > kapejod
> > >
> > > --
> > > Klaus-Peter Junghanns
> > >
> > > CEO,CTO
> > > Junghanns.NET GmbH
> > > Breite Strasse 13 - 12167 Berlin - Germany
> > > fon:+49 30 79705390
> > > fax:+49 30 79705391
> > > iaxtel: 1-700-157-8753
> > > email:  [EMAIL PROTECTED]
> > >
> > > Am Die, 2003-04-01 um 10.13 schrieb Vlasis Hatzistavrou:
> > > > Hello,
> > > >
> > > > I'd like to ask if there are any news about CE certification of the
> > > > E1 boards. I know that the T1 boards are FCC certified but I'd also
> > > > like to know what is the status for CE certification.
> > > >
> > > > Thanks for any input,
> > > > Vlasis Hatzistavrou.
> > > >
> > > > ___
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> > >
> > > ___
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> >
> >
> >
> > ___
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> >
> >
> >
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Re: [Asterisk-Users] chan_capi request

[Klaus-Peter Junghanns]

morning roy,

yes, it's possible. the settings will move into the global section
in 0.2.2.

actually there is a use for a per-device gain configuration. you might
like to have a capi device for outgoing calls to SCREAM at people
(txgain=10) ... ;-)
but i will add an option in the global section (so that the device
setting overrides the global option).

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk


Am Mit, 2003-05-28 um 14.09 schrieb Roy Sigurd Karlsbakk:
> hi all
> 
> is it hard/possible to move the following from chan_capi_pvt.h into a setting 
> (preferably global) in capi.conf?
> 
> #define AST_CAPI_NATIONAL_PREF  "0"
> #define AST_CAPI_INTERNAT_PREF  "00"
> 
> and ...
> 
> Is it hard to move or copy the txgain and rxgain to [global], either as a 
> given 'default' if nothing's set in the interfaces, or as a overall global. I 
> just can't see the use of setting gain on individual interfaces
> -- 
> Roy Sigurd Karlsbakk, Datavaktmester
> ProntoTV AS - http://www.pronto.tv/
> Tel: +47 9801 3356
> 
> Computers are like air conditioners.
> They stop working when you open Windows.
> 
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Re: [Asterisk-Users] Limit Concurrent IAX Channels

[Klaus-Peter Junghanns]

hi surajee,

some time ago martin pycko posted an example to the list
how to make use of global variables and gotoif to limit the
number of channels.
the ML archive is your friend :)

regards
kapejod
-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
Am Mon, 2003-06-02 um 17.01 schrieb Surajee Ratnayake:
> hi All,
> 
> Since the quality drops with the increased usage of IAX channels, between 2 Asterisk 
> servers,
> I want to limit the number of simultaneously used IAX channels. 
> ie basically to limit the calls between the 2 Asterisk servers, 
> can anybody pls tell me a method/hint to achieve this, i am helpless :-( 
> 
> Thanx in advance,
> Surajee

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Re: [Asterisk-Users] ISDN4Linux + Asterisk and Europe

[Klaus-Peter Junghanns]

hi kamido,


Am Die, 2003-06-03 um 12.37 schrieb [EMAIL PROTECTED]:
> Little summary what the status is:
> 
> 1. isdn4linux for real calls not just IVR forget it
> 2. capi4linux only works with cards active AND passive which do support capi
> at the moment those are eicon diva and cards from AVM
> The only passive card at the moment is the fritz card from AVM
> 
> Problems here are that the passive fritz cards have some annoying thinks like
> it's a binary library for the capi layer which is restricted for only one card 
> and no SMP box but you can circumvent this with the howto described at the 
> following site
> http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
> 
> which should enable you to use more then 1 card
> 
> however i didn't manage to get this configuration to work stable as Oliver Brandt
> reported, or I should say it's REALLY unstable at least here.
> And not only with the 2 card configuration but in any configuration
> which means 1 card,one CPU, 2 cards,one CPU and so on
> 
> so I ordered a AVM B1 PCI V 4.0   to eliminate that.
> 
> Pauline Middlelink had a suggestion which would be very nice, but i think nobody is
> able to do it at the moment . To use the zaptel driver and to implement a real 
> isdn4linux driver, because at the moment the isdn4linux is based on the modem-
> emulation of i4l and that is the reason why you have echo.

i am currently working on a zaptel BRI driver which will support the
very nice (also nicely priced) hfc-pci based cards and the multiBRI
cards. the driver will support TE and NT mode.
the reason for the echo is not only due to the high latency i4l design
(which was meant for data communication) but by todays high sensitive
mics in normal phones.

> 
> This approach would solve many problems.
> 
> You could use the Hisax cards not only on the outside like for the incoming calls 
> but also for internal isdn pbx phonesets which here in europe are cheap and have
> a lot of nice features already builtin. For this trick see isdn.jolly.de
> 
not all hisax cards do support the NT mode. iirc, only the chipsets from
CCD (hfc-pci, etc..) do.

> The echo would go away, because it is digital already and I think you could 
> count on a lower CPU usage as well.
eh?? what do you think the i4l audio is? the audio with isdn is always
digital. the term modem-emulation in i4l terms only refers to the
/dev/ttyXX stylish access from user space and the AT commands.


> 
> So maybe an interest group could get some funding together, to get more flexibility
> to the isdn side of asterisk
> 
i will gladly take every single euro!!! ;-)
> 
> hope this helps
> 
> greetings
> 
> kamido
> 

best regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
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Re: [Asterisk-Users] chan_capi syntax

[Klaus-Peter Junghanns]

Hi,

welcome to the world of digital signalling :-)

use this:

exten => _90.,1,Dial(CAPI/[msn here]:[access number])
exten => _90.,2,Wait(3)
exten => _90.,3,SendDTMF(${EXTEN:1})

regards
kapejod

P.S. inband signalling s*cks ;)

Am Don, 2003-06-19 um 14.08 schrieb WipeOut .:
> Hi,
> 
> What is the correct chan_capi dial syntax??
> 
> This is what I think it is..
> 
> exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1})
> 
> This seems to work for local numbers.. but I have an access number for cheap long 
> distance calls.. wich gets dialed and then the number I want to call is sent as DTMF 
> after a few waits (w)..
> 
> On my X100P I used the following..
> 
> exten => _9001.,1,Dial(Zap/1/[access number]www${EXTEN:1})
> 
> With chan_capi I tried..
> 
> exten => _9001.,1,Dial(CAPI/[msn here]:[access number]www${EXTEN:1})
> 
> But it doesn't seem to work.. it doesn't even dial the access number..
> 
> Any ideas??
> -- 
> __
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Re: [Asterisk-Users] best ISDN BRI solution for DID

[Klaus-Peter Junghanns]

Hi Chad,

if you are in the US you will need an ISN card that speaks NI-1.
The only CAPI cards that work with NI-1 are the Eicon Diva Server cards.
They are expensive active cards but the onboard DSP does echo
cancelation and some people already sucessfully use them in the US.

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Sam, 2003-06-21 um 04.41 schrieb Chad Sawyer:
> I want to test asterisk with its DID functions on a smaller scale without the 
> expense of a PRI.  I am in the US, and want the most reliable/stable, and feature 
> rich solution that I can have.
> 
> What ISDN card would be reccommended?  
> 
> My clec switch operator says he can/will send DID info to my BRI's if I wish.  Can 
> Asterisk, or one of its addons support this?
> 
> What BRI solution would be reccommended?  ISDN4Linux, chan_capi?  Any info is 
> appreciated, I hope to get this right the first time without having to buy more 
> hardware after the initial purchase.
> 
> Any advice from you guys with experience would be appreciated.
> 
> Chad

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Re: [Asterisk-Users] CallerID with chan_capi

[Klaus-Peter Junghanns]

Hi,

you need to order callerid from BT and pay for it.

regards
kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Don, 2003-06-26 um 15.42 schrieb WipeOut .:
> Is there anything special that needs to be done to get the CallerID to work with 
> chan_capi??
> 
> When a call come in the display on the phone shows things like "€£".. have I set 
> something up wrong?
> 
> Thanks
> -- 
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Re: [Asterisk-Users] Conference calls

[Klaus-Peter Junghanns]

Hi,

if you dont have usb-uhci you can also use your realtime clock
to generate zaptel timing. Make sure you dont have rtc support
compiled into your kernel and grab zaprtc from:
http://www.junghanns.net/asterisk

regards
kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Die, 2003-07-01 um 16.29 schrieb Martin Pycko:
> You need to look at "show application meetme" in the asterisk CLI
> but for it to work you need to have some kind of zaptel hardware or
> emulate it with zttdummy (but for that you need to have usb-uhci like USB
> controller)
> 
> and then
> 
> exten => 1000,1,Meetme,1000
> 
> Martin
> 
> On Tue, 1 Jul 2003, Serge Mankovski wrote:
> 
> > Hi
> > I want to set up * as a conference bridge. I would like to be able to
> > conference is SIP calls (up to 12)
> >
> > I am looking through all available documentation for * to get info on how it
> > is done. No luck so far.
> >
> > Can somebody direct me to the info in this subject?
> >
> > Thank you
> > Serge
> >
> > _
> > Protect your PC - get McAfee.com VirusScan Online
> > http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963
> >
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Re: [Asterisk-Users] *--IAX--* problems. (chan_capi problem)

[Klaus-Peter Junghanns]

Hi Emanuele,

update to 0.2.4.

best regards

kapejod

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fon:+49 30 79705392
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Am Die, 2003-07-22 um 21.36 schrieb Emanuele Pucciarelli:
> Il mar, 2003-07-22 alle 15:41, WipeOut . ha scritto:
> > Ok.. I have done some more digging and the problem seems to be caused by chan_capi 
> > not detecting that the call has been answered.. I downgraded chan_capi from 0.2.3b 
> > to 0.2.2 and the system is working fine..
> 
> I have tried to tackle that issue today.  It seems that the problem of
> answers not being detected can be solved by uncommenting lines 1712 and
> 1718 in chan_capi.c (0.2.3b), but I haven't yet got a chance to test
> what REALLY happens then. :)
> 
> It just seems to work, but further surprises might be just round the
> corner...
> 
> --
> Emanuele
> 
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Re: [Asterisk-Users] Asterisk IVR and Hicom 300

[Klaus-Peter Junghanns]

Hi,

if your Hicom offers internal S0 (isdn) to connect isdn phones
then you can just take a passive AVM Fritz Card and connect it
to the Hicom.

regards

kapejod
-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mit, 2003-07-23 um 13.55 schrieb Dr. Andreas Moroder:
> Hello,
> 
> in our public hospital we have 3 hicom 300. We now would like to have IVR.
> Is it possible to use Asterisk togheter with the hicom as IVR ?
> What hardware do we need to connect the Asterisk server with the Hicom ? is ISDN
> enough ?
> 
> Regards
> Andreas Moroder
> 
> --
>   Dr. Andreas Moroder
> Leiter der Informatikabteilung - Direttore della ripartizione informatica
> Sanitätsbetrieb Brixen - Azienda Sanitaria di Bressanone
>   Tel. 0472 812085   Fax 0472 812089
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Re: [Asterisk-Users] Configuration sample for isdn4linux?

[Klaus-Peter Junghanns]

Hi Holger,

the AVM-A1 (fritz card) has capi4linux drivers (ftp.avmd.de/cardware)
which may already be included in your distro.

get chan_capi at http://www.junghanns.net/asterisk/ and read the
README and INSTALL file.

regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
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Am Fre, 2003-07-25 um 11.55 schrieb Holger Wirtz:
> Hi,
> 
> we want to use asterisk as a replacement for our current pbx but before
> changeing we want to make some tests.
> 
> Therefor I have a PC with an AVM-A1 ISDN4Linux hardware and inside the
> network a cisco 7960 Ip Phone (SIP) and two TIPTEL innovaphone 200 (H.323) 
> and some softpgones like kphone, netmeeting, ...
> 
> To learn something about the configuration of asterisk I want to create
> a "small" config where all my clients can call each other AND ca use the
> two B-Channels of my isdn4linux-card.
> 
> I read the asterisk handbook and I tried hard to use the installed
> config files but I have real problems to understand  the examples in the
> handbook or the configs because they are "for all reasons".
> 
> Does anyone have such a configuration in use and can send me the
> configs? This would be very great! I hope to learn more by using a
> working config and try to get it to work with my environment.
> 
> Thanx, Holger
> 
> -- 
> #   ##  ##   Holger Wirtz Phone : (+49 30) 884299-40
> ##  ## ##   ### ##   DFN-Verein   Fax   : (+49 30) 884299-70
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Re: [Asterisk-Users] executing an agi script after a successful Dial

[Klaus-Peter Junghanns]

Hi Dan,

no wonder. when the h extension is called the channel (including all
the channel variables you want to read with get_var) is gone. pass the
channel variables you need to acces as an argument to the agi script,
e.g.: exten => h,1,AGI(myagi.agi,${EXTEN} ${CALLERIDNUM})

regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Sam, 2003-07-26 um 01.28 schrieb Dan Fernandez:
> John
> 
> Thanks for the response.  This seems to be what I am looking. However, I
> have discovered a problem with a simple perl script triggered from the h
> extension.
> 
> I am using perl-Asterisk and if I call the script from any extension in
> works fine. However, if I call the same script from h the get_variable and
> verbose functions don´t work anymore.
> 
> Rgds
> Dan
> - Original Message -
> From: "John Todd" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, July 23, 2003 8:20 PM
> Subject: Re: [Asterisk-Users] executing an agi script after a successful
> Dial
> 
> 
> > >On Wednesday 23 July 2003 05:04 pm, Dan Fernandez wrote:
> > >>  I would like to run an agi script (to calculate the cost of a long
> > >>  distance or international call) right after I execute a Dial app.
> > >>  Can this be configured in extensions.conf? It seems the entries
> > >
> > >It cannot.  If the Dial app succeeds in getting a connected channel,
> > >it will ALWAYS return -1, which signals a hangup to Asterisk.  The
> > >only time Dial will ever return control to the dialplan is if either
> > >the channel is not available or if the channel does not get connected.
> >
> > Hmm... I'm not so sure about what the question was, and if perhaps
> > there is some confusion about what is desired here.  In my example
> > configs, I use the "h" extension to clean up call recording after
> > Dial has terminated.  Seems to work for me, but perhaps it's not
> > supposed to work.  :)
> >
> > Dan - try putting your routines in an extension called "h".  This may
> > get executed after Dial terminates normally or abnormally.
> >
> > JT
> >
> >
> > >  > right after a Dial app get executed only if the Dial app was
> > >>  executed unsucessfully. Would I have to execute the dial app from
> > >>  the agi script?
> > >
> > >No, again, the Dial app won't return control to the AGI script until
> > >after the call is complete.  You're pretty much going to have to do
> > >whatever you want to do prior to executing Dial or after the call is
> > >complete.  Of course, you could create a separate thread which
> > >runs parallel to the channel thread and does various monitoring
> > >tasks, but that would require some C programming skills.
> > >
> > >-Tilghman
> > >
> > >___
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Re: AW: [Asterisk-Users] * behind ISDN pbx - Forwarding toextensions with in primary pbx

[Klaus-Peter Junghanns]

Morning Peer,


Am Mon, 2003-07-28 um 09.32 schrieb Peer Oliver schmidt:
> Peer Oliver schmidt wrote:
> > Hi Andreas,
> > 
> >>> I have asterisk behind my primary PBX connected via ISDN (chan_capi).
> >>> Calling out and calling in works just fine, however I can't connect to
> >>> my primary pbxs' extensions.
> > 
> > 
> >> at my site it is working exactly as you wrote in your 1st example. How is
> >> your PBX setup? I remember that there is a way to set a pbx to spontanic
> >> trunk access. At least my Agfeo has got such a setup possibility. Try to
> >> switch this off for your ISDN Card.
> > 
> > 
> > Hmm, it is turned off. However still no luck. Are you using CAPI as 
> > well? What version * and the capi_chan?
> 
> I did some more digging. If I understand the following log correct, 
> strange things happen:
> 
>  -- Executing Wait("SIP/pos-d2db", "1") in new stack
>  -- Executing Dial("SIP/pos-d2db", "CAPI/30:25|20") in new stack
>  -- data = 30:25
>  -- capi request omsn = 30
>== found capi with omsn = 30
>== CAPI Call CAPI[contr2/30]/24   == CAPI Call CAPI[contr2/30]/24 
>   -- Called 30:25
>== received CONNECT_CONF PLCI = 0x102 INFO = 0
>  -- CAPI[contr2/30]/24 is making progress passing it to SIP/pos-d2db
> > activehangingup
>== DISCONNECT_IND PLCI=0x102 REASON=0x349c
>== No one is available to answer at this time
> 
> 
> The Executing Dial "CAPI/30:25|20" should call extension 25 for 20 
> seconds using the outgoing channel with the MSN 30, shouldn't it? What I 
> don't understand is
> 
> CAPI Call CAPI[contr2/30]/24
> 
> Does it try to call extension 24 now?!?

no, it doesnt. the "/24" is just for making sure that the channel name
is unique (you will notice that the 24 gets incremented with every call)

your pbx complains with REASON=0x349C which means "Invalid number
format".

you need to find a way to turn off "spontane amtsholung" or try to
prefix internal numbers with "**", this works for siemens style pbxes.

> 
> Anyone able to shed some light on this?
> 
> TIA
> rgds
> pos
> 
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regards

kapejod

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Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
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Re: [Asterisk-Users] Problems with two B channels

[Klaus-Peter Junghanns]

Hi creator of the first H323 channel ;-)

does the B1 get its own irq? did you try different versions of
AVM's capi drivers?

regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Mon, 2003-07-28 um 15.00 schrieb Michael Manousos:
> 
> Hello all,
> 
> I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA)
> on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one
> B channel. But when I open a second B chan, the sound is choppy,
> with too long gaps, and the CPU load is too high (~50%).
> On the Asterisk's console I get these messages:
> 
>  -- Executing Dial("H323:4478", "CAPI/7810:[MY_NUMBER_HERE]") in new 
> stack
>  -- Called 7810:[MY_NUMBER_HERE]
>  -- CAPI[contr1/7810] is making progress passing it to H323:4478
>  -- CAPI[contr1/7810] is ringing
> kcapi: appl 1 ncci 0x20201 up
>  -- CAPI[contr1/7810] answered H323:4478
> ERROR[344086]: File chan_capi.c, Line 900 (capi_write): error sending 
> DATA_B3_REQ (error=0x1103, datalen=160)
> ERROR[344086]: File chan_capi.c, Line 900 (capi_write): error sending 
> DATA_B3_REQ (error=0x1103, datalen=160)
> ERROR[262161]: File chan_capi.c, Line 900 (capi_write): error sending 
> DATA_B3_REQ (error=0x1103, datalen=160)
> ERROR[344086]: File chan_capi.c, Line 900 (capi_write): error sending 
> DATA_B3_REQ (error=0x1103, datalen=160)
> ERROR[262161]: File chan_capi.c, Line 900 (capi_write): error sending 
> DATA_B3_REQ (error=0x1103, datalen=160)
> 
> Lots of these messages!
> 
> I have tried both 0.2.2 and 0.2.4a versions of CAPI driver.
> 
> Thanks,
> Michael.
> 
> 
> 
> 
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Re: [Asterisk-Users] Problems with two B channels

[Klaus-Peter Junghanns]

Hi,

did you try the original AVM drivers from ftp.avm.de/cardware?
what else is running on the box? loadavg?

230 irqs/sec sounds a lot, even for 2 B channels, but i dont
have any experience with the AVM pcmcia cards.

regards

kapejod

Am Mon, 2003-07-28 um 17.08 schrieb Michael Manousos:
> Klaus-Peter Junghanns wrote:
> > Hi creator of the first H323 channel ;-)
> 
> Hi creator of the only CAPI channel driver :))
> 
> > 
> > does the B1 get its own irq? did you try different versions of
> > AVM's capi drivers?
> 
> Yes, the B1 gets its own IRQ and, yes, I did try different
> AVM firmwares (ftp://ftp.in-berlin.de/pub/capi4linux/firmware).
> An important notice. The device gets aprox. 230 interrupts/sec
> on a StrongARM 1110 based platform (200MHz).
> Is this normal?
> 
> > 
> > regards
> > 
> > kapejod
> > 
> 
> Thanks,
> Michael.
> 
> 
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Re: [Asterisk-Users] CAPI & CLID

[Klaus-Peter Junghanns]

Hi Stuart,

you have to order CLIP for your BRI from BT and pay
them for it.

regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Die, 2003-07-29 um 22.43 schrieb Stuart Hirst:
> I have an AVM Fritz card using CAPI which seems to work quite well apart
> from the CLID is not being captured correctly.
> 
> On my SNOM 200 the CLID displays odd characters and in CAPI debug the
> CLID reported is "£".
> 
> Anyone any ideas ?
> 
> Rgds,
> 
> Stuart
> 
> 
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Re: [Asterisk-Users] chan_capi: Hanging channels - again

[Klaus-Peter Junghanns]

Hi Roy,

always use latest chan_capi. the bug is fixed in 0.2.4a.
today 0.2.4b is online which fixes some issues with sending
dtmf and a small enhancement to capiECT.

capi on!

regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Die, 2003-08-05 um 15.41 schrieb Roy Sigurd Karlsbakk:
> hi all
> 
> sad news
> 
> those channels are hanging again.
> 
> Anything more I can do to troubleshoot this?
> 
> *CLI> show channels
> Channel  (ContextExtensionPri )   State Appl. Data
> CAPI[contr2/22545079]/57  (ola22545079 1   )  Up Bridged Call  
> SIP/ola-ac25
>SIP/ola-ac25  (default51676840 1   )  Up Dial  
> CAPI/22545079:b51676840|300|T
> CAPI[contr2/22545066]/55  (torgeir22545066 1   )Down (None)
> (None)
> CAPI[contr2/22545066]/54  (torgeir22545066 1   )Down (None)
> (None)
> 4 active channel(s)
> *CLI>
> 
> -- 
> Roy Sigurd Karlsbakk, Datavaktmester
> ProntoTV AS - http://www.pronto.tv/
> Tel: +47 2254 5070 (work)
>  +47 9801 3356 (mobile)
> 
> Computers are like air conditioners.
> They stop working when you open Windows.
> 
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Re: [Asterisk-Users] chan_capi: Hanging channels - again

[Klaus-Peter Junghanns]

http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.4b.tar.gz

the downloads dir is browseable, but i probably should update the
 website a bit

regards

kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
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Am Don, 2003-08-07 um 12.42 schrieb Armand A. Verstappen:
> Hi Klaus-Peter,
> 
> On Wed, 2003-08-06 at 12:33, Klaus-Peter Junghanns wrote:
> > always use latest chan_capi. the bug is fixed in 0.2.4a.
> > today 0.2.4b is online which fixes some issues with sending
> > dtmf and a small enhancement to capiECT.
> 
> I checked the site, but can't find the 0.2.4b version. The sidebar menu
> offers 0.2.4a, and http://www.junghanns.net/asterisk/page1.html offers
> 0.2.2 (under download lates version _here_).
> 
> wkr,
> 
> -- 
> Envida http://www.envida.net/
> Armand A. Verstappen   Graadt van Roggenweg 328
> [EMAIL PROTECTED]   3531 AH Utrecht
> tel: +31 (0)30 298 2255Postbus 19127
> fax: +31 (0)30 298 21113501 DC Utrecht

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Re: [Asterisk-Users] avm fritz pci

[Klaus-Peter Junghanns]

hi Marian,

you cannot use the AVM Fritz in P2P mode (at least not with
the capi drivers), maybe I4L will work (hisaxctrl 
7 1) IIRC.

regards

kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
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Am Mon, 2003-08-11 um 18.03 schrieb Marian Danisek:
> hello,
> 
> does anybody know how to setup avm fritz pci card in p2p mode ?
> 
> regards marian
> 
> -- 
> SUNTEQ s. r. o.
> Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic
> Tel: +421-46-5430 754 # Fax: +421-46-5439 144
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> 
> A mind is like a parachute... it only works when it's open.
> 
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Re: [Asterisk-Users] Chan_Capi questions??

[Klaus-Peter Junghanns]

Hi WipeOut,

softdtmf=1 will use asterisk's dsp functions to detect and
generate the DTMF tones. softdtmf=0 will use your capi controller
to do the detection/generation. unless you have an active
card i would suggest to use softdtmf=1.

early B3 connects will let you hear the inband call progress
that your telco's switch sends you (ring indication,).
the "b" enables all inband call messages (including those
"your call cannot be completed as dialed", busy ). this may
not be suitable for all type of applications, therefore there
is the "B" option which enables early B3 connects on success.
That means you will get ring indication or you can get the
dialtone from your telco by dialing an empty number.
In a normal pbx application the "b" option makes sense.

regards

kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Sam, 2003-08-09 um 14.54 schrieb WipeOut .:
> I have been using chan_capi and a Fritz AVM (passive) card for a little while now.. 
> So far so good.. There are still a couple of things I am not totally sure about..
> 
> First..
> 
> What is the difference between "softdtmf=0" and "softdtmf=1"??
> Which is the preffered/recommended option??
> 
> Second..
> 
> What is Early B3 all about??
> Should I be using it in my dial string??
> Which is the preffered/recommended option "b" or "B"??
> 
> Thanks..
> -- 
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Re: [Asterisk-Users] chan_capi compile errors with latest CVS

[Klaus-Peter Junghanns]

Hi,

chan_capi 0.2.4d fixes the problem.

regards

kapejod

-- 
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CEO,CTO
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Am Son, 2003-08-17 um 15.32 schrieb Michiel Betel:
> Did something change in lock.h lately? I get all kind of ast_mutex errors
> when trying to compile chan capi 0.24c with the latest asterisk code
>  
>  
>  
> 
> Betel Consultancy
> Abelenlaan 19 T: +31 20 640 3018
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> The Netherlands W:  <http://www.betel.nl/> www.betel.nl
> 
> 
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Re: [Asterisk-Users] fatal embrace control in menus ?

[Klaus-Peter Junghanns]

Morning,

this is just too simple ;-)
Try this if you want to loop 3 times:

exten => s,1,Answer()
exten => s,2,SetVar(LOOPS=0)
exten => s,3,Background(MenuSound)
exten => s,4,SetVar(LOOPS=$[${LOOPS} + 1])
exten => s,5,GotoIf($[${LOOPS} < 3]?3:6)
exten => s,6,Hangup()

regards

kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
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fon:+49 30 79705392
fax:+49 30 79705391
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Am Son, 2003-08-24 um 11.08 schrieb Todd Lieberman:
> I just repeat line 2, e.g.
> 
> exten => s,1,Play(Greeting)
> exten => s,2,Background(Menu Sound)
> exten => s,3,Background(Menu Sound)
> 
> 
> 
> 
> 
> John Brown wrote:
> 
> >Hi, 
> >
> > So how does one keep this from happening.
> >
> >(note, the following is fake code)
> >
> >exten => s,1,Play(Greeting)
> >exten => s,2,Background(Menu Sound)
> >exten => s,3,goto(s,2)
> >
> >How does one control the number of times s,3 gets executed??
> >
> >The issue is this.
> >
> >1. Call comes in via PSTN, * answers the call, starts to play
> >   things, caller hangs up.  Menu loops for quite some time :)
> >
> >2. Call comes in via PSTN, * answers the call, start to play things
> >   caller does nothing on the first pass thru the menu.
> >
> >   I want the menu to repeat X times and then disconnect if no
> >   action from the claller.
> >
> >
> >samples of extensions.conf would be really appreciated
> >
> >
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Re: [Asterisk-Users] I4L CallerID not working

[Klaus-Peter Junghanns]

Hi Adam,

the telco is not sending CallerID to your line. Otherwise you would find
a "Calling Party Number IE" in the SETUP message.

regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Mon, 2003-08-25 um 13.19 schrieb Adam Goryachev:
> Can anyone work out why my callerid doesn't work on my isdn4Linux with
> asterisk (or without asterisk for that matter)...
> 
> This used to work fine, and I am quite confident that the telco is sending
> callerid information (because they always do on all ISDN lines standard,
> only extra cost on POTS lines).
> 
> This is the information from dmesg, whether asterisk is running or not:
> isdn_net: Incoming call without OAD, assuming '0'
> isdn_net: call from 0 -> 0  ignored
> 
> where  is my local phone number without area code
> I turned on the maximum isdn debugging, and here is the output when I dial
> in, at this time, asterisk was not running...
> 
> 29:42.76 Card1 tiger: i1 2 80
> 29:42.76 Card1 ISAC interrupt 80
> 29:42.76 Card1 isac_empty_fifo cnt 4 02 81 01 3B
> 29:42.76 Card1 <- PH_DATA: RR[1](nr 29)C (sapi 0, tei 64)
> HEX: 02 81 01 3B
> 29:42.76 frame network->user SFrame with tei 64
> 29:42.78 Card1 tiger: i1 2 80
> 29:42.78 Card1 ISAC interrupt 80
> 29:42.78 Card1 isac_empty_fifo cnt 4 00 81 01 3B
> 29:42.78 Card1 <- PH_DATA: RR[1](nr 29)R (sapi 0, tei 64)
> HEX: 00 81 01 3B
> 29:42.78 frame network->user SFrame with tei 64
> 29:45.87 Card1 tiger: i1 2 80
> 29:45.87 Card1 ISAC interrupt 80
> 29:45.87 Card1 isac_empty_fifo cnt 31 02 FF 03 08 01 03 05 A1 04 03 80 90 A3
> 18 01 8A 70 09 C1 39 36 39 34 31 31 30 30 7D 02 91 81
> 29:45.87 Card1 <- PH_DATA: UI[0]C (sapi 0, tei 127)
> HEX: 02 FF 03 08 01 03 05 A1 04 03 80 90 A3 18 01 8A 70 09 C1 39 36 39 34 31
> 31 30 30 7D 02 91 81
> 29:45.87 frame network->user broadcast
> callref 3 caller size 28 message type SETUP
>   Sending complete
>   Bearer capability
> octet 3  1000
> octet 4  1001
> octet 5  10100011
>   Channel identification
> octet 3 10001010
>   Called party number
> octet 3 1101
> number digits 96941100
>   High layer compatibility
> octet 3  10010001
> octet 4  1001
> 29:45.87 DCh Q.921 State ST_L2_1 Event EV_L2_UI
> 29:45.87 Ch0 callc State ST_NULL Event EV_SETUP_IND
> 29:45.87 Ch0 callc ChangeState ST_IN_WAIT_LL
> 29:45.87 Ch0 HL->LL STAT_ICALL
> 29:45.87 Ch0 LL->HL statcallb ret=0
> 29:45.87 L3DC State ST_L3_LC_REL Event EV_RELEASE_REQ no routine
> 29:45.87 Ch0 callc ChangeState ST_NULL
> 29:49.92 Card1 tiger: i1 2 80
> 29:49.92 Card1 ISAC interrupt 80
> 29:49.92 Card1 isac_empty_fifo cnt 31 02 FF 03 08 01 03 05 A1 04 03 80 90 A3
> 18 01 8A 70 09 C1 39 36 39 34 31 31 30 30 7D 02 91 81
> 29:49.92 Card1 <- PH_DATA: UI[0]C (sapi 0, tei 127)
> HEX: 02 FF 03 08 01 03 05 A1 04 03 80 90 A3 18 01 8A 70 09 C1 39 36 39 34 31
> 31 30 30 7D 02 91 81
> 29:49.92 frame network->user broadcast
> callref 3 caller size 28 message type SETUP
>   Sending complete
>   Bearer capability
> octet 3  1000
> octet 4  1001
> octet 5  10100011
>   Channel identification
> octet 3 10001010
>   Called party number
> octet 3 1101
> number digits 96941100
>   High layer compatibility
> octet 3  10010001
> octet 4  1001
> 29:49.92 DCh Q.921 State ST_L2_1 Event EV_L2_UI
> 29:49.92 Ch0 callc State ST_NULL Event EV_SETUP_IND
> 29:49.92 Ch0 callc ChangeState ST_IN_WAIT_LL
> 29:49.92 Ch0 HL->LL STAT_ICALL
> 29:49.92 Ch0 LL->HL statcallb ret=0
> 29:49.92 L3DC State ST_L3_LC_REL Event EV_RELEASE_REQ no routine
> 29:49.92 Ch0 callc ChangeState ST_NULL
> 29:52.80 Card1 tiger: i1 2 80
> 29:52.80 Card1 ISAC interrupt 80
> 29:52.80 Card1 isac_empty_fifo cnt 4 00 81 01 3B
> 29:52.80 Card1 <- PH_DATA: RR[1](nr 29)R (sapi 0, tei 64)
> HEX: 00 81 01 3B
> 29:52.80 frame network->user SFrame with tei 64
> 30:02.83 Card1 tiger: i1 2 80
> 30:02.83 Card1 ISAC interrupt 80
> 30:02.83 Card1 isac_empty_fifo cnt 4 00 81 01 3B
> 30:02.83 Card1 <- PH_DATA: RR[1](nr 29)R (sapi 0, tei 64)
> HEX: 00 81 01 3B
> 30:02.83 frame network->user SFrame with tei 64
> 
> Any hints/clues would be very handy, as it is kind of annoying to have
> callerid available but not be able to make use of it...
> 
> Regards,
> Adam
> 
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Re: [Asterisk-Users] te410p with serial console fails with error:TE410P: Double/missed interrupt detected

[Klaus-Peter Junghanns]

Hi,

read /usr/src/linux/Documentation/networking/netlogging.txt

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Mon, 2003-08-25 um 16.50 schrieb Mike Ciholas:
> 
> On Mon, 25 Aug 2003, Gavin Hollinger wrote:
> 
> > Thanks Mark, Sorry for all the questions, I am treading on
> > un-familiar ground here.  Can anyone think of a way to adjust
> > how long the serial console holds the interrupt when the kernel
> > sends a message?  Anyone know what the purpose of this behavior
> > is?
> 
> Here are some uneducated thoughts:
> 
> 1. Make serial port polled instead of interrupt driven?  Majority 
> of traffic is output that will be buffered.
> 
> 2. Increase serial port buffer size?  Perhaps problem is so much 
> serial output that buffer fills up and blocks.
> 
> 3. Make serial port faster?  Would help drain buffer quicker.  
> 115200 baud is routine, I've seen 460800 baud being used for 
> embedded Linux on short cables.
> 
> 4. Is there a way to have a remote console via ethernet?  Won't 
> help with BIOS settings, but would remove serial port issues.
> 
> 5. Use different serial port hardware (USB serial dongle, PCI
> multiport card, etc) for console.  Switch serial cable to do BIOS
> settings if need be.
> 
> What BIOS do you have that is serial configurable?  I'm always on 
> the lookout for that.
> 
> -- 
> Mike Ciholas(812) 476-2721 voice
> CIHOLAS Enterprises (812) 476-2881 fax
> 2626 Kotter Ave, Unit D [EMAIL PROTECTED]
> Evansville, IN 47715http://www.ciholas.com
> 
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Re: [Asterisk-Users] te410p with serial console fails with error:TE410P: Double/missed interrupt detected

[Klaus-Peter Junghanns]

Hi Gavin,

sorry, i was assuming that everybody uses SuSE kernels (which have
a lot of useful stuff already built in)... :-)

here is the URL for the netconsole patches:
http://www.kernel.org/pub/linux/people/mingo/netconsole-patches


best regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Mon, 2003-09-01 um 18.08 schrieb Gavin Hollinger:
> >> read /usr/src/linux/Documentation/networking/netlogging.txt
> > It doesn't exist there in 2.4.22-rc4.
> 
> After searching the net, I was able to find many references to this
> document but no links to the document itself.  None of my Linux systems
> had it either.  I would appreciate if someone could post it or point me to
> it.
> 
> The concept of a "network console" from kapejod was most intriguing for
> several reasons.  Especially higher speed in dumping error messages. 
> However, due to lack of know how, I ended up creating 2 entries in GRUB
> one with and one without the serial console.  Because we do almost
> everything with ssh anyway, that will get us by for now.  When we really
> need the serial console to fix something, we can reboot, and use it but
> not asterisk.
> 
> Thanks in no particular order to the following for your input on this
> thread.  You are most helpful:
> 
> Mark Spencer
> Richard Scobie
> Mike Ciholas
> Ray Burkholder
> Dave Cotton
> Klaus-Peter Junghanns (kapejod)
> 
> 
> Gavin
> 
> 
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Re: [Asterisk-Users] Stuck On ISDN

[Klaus-Peter Junghanns]

Hi Olle,

the cheapes CAPI card is the passive AVM Fritz Card PCI. It's nice
to start playing, but for a production system (isdn pbx) i can only
recommend the active Eicon Diva Server cards (which support EC).

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Die, 2003-09-02 um 21.36 schrieb Olle E. Johansson:
> The devices that support CAPI seems much more expensive than ISDN Cards
> with I4L support. What's the least expensive ISDN card with support for CAPI?
> 
> I finally got I4L working and the sound quality could be better, but it
> works for experimental systems. Since I do not _really_ understand how
> I got it working from a state with no reaction at all I can't explain
> or assist, just send my config file on request.
> 
> The manual says nothing on the subject on I4L support and there's
> nothing to find on the net by googling... Everyone points to
> capi and, back to the start of my reply, it seems expensive for personal
> use...
> 
> /Olle
> 
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Re: [Asterisk-Users] SIP to CAPI problem

[Klaus-Peter Junghanns]

Hi Ben,

dont use the 'r' option in your dialstring. chan_capi will
tell asterisk when the remote end is ringing , no need to fake
that. also the 't' option may not be what you want, or do you
want to allow the called party to transfer themselves inside of
your asterisk? ;-)

best regards

kapejod

-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mit, 2003-09-03 um 17.57 schrieb bbb bben:
> Hi,
> 
> I'm using * with SIP in our LAN and CAPI for external
> phone calls.
> Everything works fine (SIP to SIP calls are ok and
> external incomming calls are ok too (CAPI to SIP))
> exept the CAPI to SIP communication.
> When I try to make an external call from a SIP client,
> the (ext) phone rings (while the SIP client display
> "trying...") but when I hangup I don't get any
> communication and the SIP client stay in its
> "trying..." state.
> 
> Here is what the * consol shows:
> 
> -- Executing Dial("SIP/1000-ce58",
> "CAPI/32:076XX|tr") in new stack
> -- creating pipe for PLCI=-1
> -- Called 32:076XX
> -- CAPI Hangingup
> -- Setting up echo canceller (PLCI=0x101,
> function=1, options=2, tail=64)
> -- Echo canceller successfully set up (PLCI=0x101)
> -- removed pipe for PLCI = 0x201
> 
> nothing more. What is happening? Ideas?
> Can anybody help?`
> 
> =Ben=
> 
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Re: [Asterisk-Users] Sound error during launch

[Klaus-Peter Junghanns]

Hi Martin,

i had the same problem (and also asterisk wont share your sound device
with anything else). try my chan_oss patch:
www.junghanns.net/asterisk/downloads

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Son, 2003-09-07 um 21.59 schrieb marrandy:
> Hello.
> 
> Although I can hear the demo etc. now, I notice during asterisk launch I get 
> :-
> [chan_oss.so] => (OSS Console Channel Driver)
>   == Console is full duplex
>   == Registered channel type 'Console' (OSS Console Channel Driver)
>   == Parsing '/etc/asterisk/oss.conf': Found
> WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound 
> device: Resource temporarily unavailable
> 
> I'm running mandrake 9.1 with the ALSA sound driver.  
> In the control center | Hardware | HardDrake | soundcard | help
> 
> it suggest this is preferred.
> 
> Is that warning important ?
> 
> How do I correct it ?
> 
> Regards...Martin
> -- 
> It is better to be bow-legged than no-legged.
> 
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Re: [Asterisk-Users] Sound error during launch

[Klaus-Peter Junghanns]

Hi,

maybe it's "a deeply silly thing to do". But it makes chan_oss
usable for me (chan_alsa never worked for me). I do all of my calls
(voip and capi) with chan_oss without ever having had any audio
problems.
Withouth that patch every "console beep" will make * hang up the 
channel, which is really annoying...

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mon, 2003-09-08 um 11.17 schrieb Alastair Maw:
> Klaus-Peter Junghanns wrote:
> 
> > i had the same problem (and also asterisk wont share your sound device
> > with anything else). try my chan_oss patch:
> > www.junghanns.net/asterisk/downloads
> 
> Please don't use this patch - commenting out log messages is a deeply 
> silly thing to do.
> 
> If you want to use ALSA rather than OSS, put the following in your 
> modules.conf file:
> 
> noload => chan_oss.so
> load => chan_alsa.so
> 
> Note that if you don't have a sound card at all, you should noload both 
> of these.
> 
> -- 
> Alastair Maw <[EMAIL PROTECTED]>
> MX Telecom - Systems Analyst
> http://www.mxtelecom.com
> 
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Re: [Asterisk-Users] CAPI silence detection

[Klaus-Peter Junghanns]

hi,

there is NO option to disable silence detection for
chan_capi, because chan_capi DOES NOT support silence 
detection. :-)

regards

kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mit, 2003-09-10 um 10.28 schrieb Rattana BIV:
> 
> Hi,
> 
> Where can I disable silence detection with chan_capi ?
> Is there option in capi.conf ?
> 
> 
> Regards
> Rattana

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Re: [Asterisk-Users] CAPI silence detection

[Klaus-Peter Junghanns]

do you see DATA_B3_REQ errors on the * console? if yes, then
netmeeting has a bad timing (or no timing?) for sending audio
or your network produces that jitter. chan_capi needs to get
the outgoing audio in a rather strict timing (there is no 
possibility to use a jitterbuffer on a Bchannel! 64 kbits is
64 kbits, if something comes too late it will create jitter).

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mit, 2003-09-10 um 10.59 schrieb Rattana BIV:
> OK
> 
> When I call Netmeeting by my phone. I have silence (the sound is choppy) in
> my phone but not with netmeeting.
> 
> And I don't know why ?
> How can I set it ?
> If chan_capi not support silence detection perhaps asterisk do it ...
> any tips ...
> 
> 
> Thanks
> Rattana
> 
> - Original Message -
> From: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, September 10, 2003 10:26 AM
> Subject: Re: [Asterisk-Users] CAPI silence detection
> 
> 
> > hi,
> >
> > there is NO option to disable silence detection for
> > chan_capi, because chan_capi DOES NOT support silence
> > detection. :-)
> >
> > regards
> >
> > kapejod
> >
> > --
> > Klaus-Peter Junghanns
> >
> > CEO,CTO
> > Junghanns.NET GmbH
> > Breite Strasse 13 - 12167 Berlin - Germany
> > fon: +49 30 79705392
> > fax: +49 30 79705391
> > iaxtel: 1-700-157-8753
> > email: [EMAIL PROTECTED]
> > http://www.junghanns.net/asterisk
> >
> > Am Mit, 2003-09-10 um 10.28 schrieb Rattana BIV:
> > >
> > > Hi,
> > >
> > > Where can I disable silence detection with chan_capi ?
> > > Is there option in capi.conf ?
> > >
> > >
> > > Regards
> > > Rattana
> >
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Re: [Asterisk-Users] chan_capi

[Klaus-Peter Junghanns]

Hi Alexander,

you will probably find no Telco that uses ECT in their network,
app_capiECT works fine on most isdn PBXes though.
Concerning app_capiCD...just give it a try, but of course your
telco will have to support CD. "does not rely on sservice CD" 
refers to the way it is implemented in chan_capi (it uses an
INFO_REQ instead of a FACILITY_REQUEST).

My cheap isdn pbx with an internal S0 bus happily supports ECT
but not CD (so i have 1 capi card on the internal S0 bus for ECT 
and 1 card on the external line for CD).

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Son, 2003-09-14 um 17.57 schrieb Alexander Noack:
> Hi chan_capi users,
> 
> this thing is awesome, no delays like in modem_i4l!
> Plus, it got those nice ISDN features.
> 
> Here's my question:
> Does my service provider (Deutsche Telekom) have to provide me with
> these Services (CD, ECT)?
> (the Readme in 0.2.5 says "does not relay on service CD")
> 
> I know, that I don't have CFU,CFNR,CFBS (which I would have to order
> seperately).
> How likely would it be that an ISDN-PBX with an internal S0 supports
> CD,ECT services, if the line to my telco doesn't?
> 
> 
> Here's what I get:
> I get REASON=0x3400 as a disconnect indicator when trying out CD.
> (calling in on one ISDN line from my mobile number deflecting to my
> other ISDN line; as laid out in the Readme)
> 
> I get REASON=0x3490 as a disconnect indicator for the onHOLD party,
> trying ECT.
> capiECT says the call has been answered (I picked up the ringing phone
> where the call was being transferred to), but then I get a warning
> "file.c Line 823 (ast_waitstream): Unexpected control subclass '14'".
> This nice lady is reading the digits of the calling party, then nothing
> happens until a timeout occurrs.
> 
> 
> I am not a C programmer but I know how to read some source code ... I'd
> gladly help where I can!
> 
> Any suggestions are greatly appreciated.
> Alex
> 
> - Alexander Noack
> - Vitus-Bering-Str. 3
> - 18106 Rostock
> - Germany
> -
> -Tel: +49 (381) 12839893
> -Fax: +49 (381) 12839901
> -Web: www.gaussdivision.de
> 
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Re: [Asterisk-Users] ISDN BRI active adapters with NT mode - any alternatives ?

[Klaus-Peter Junghanns]

Hi,

i dont think that the Eicon Diva Server 4BRI's NT mode feature will
work with linux/capi. I think the feature in the driver is for their
PRI cards (where everything is always P2P). i may be wrong, though.
We are working on an alternative, a passive multiport ISDN card that
supports TE and NT mode with zaptel drivers for asterisk.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
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Am Mon, 2003-09-15 um 11.42 schrieb Jean-Marc V. Liotier:
> Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking
> for a CAPI compatible active BRI adapter capable of NT mode. I have
> found that the 'Eicon Diva Server 4BRI' fulfills theses requirements.
> However, at EUR 1387 for four BRI interfaces, it does not come cheap.
> Are there other users of similar hardware that can point me to
> alternatives ?
> 
> 
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Re: [Asterisk-Users] ISDN BRI active adapters with NT mode - any alternatives ?

[Klaus-Peter Junghanns]

Hi,

the AVM cards are not capable of NT mode. They also dont support
echo cancellation like the Eicons do, plus the C4 is more expensive
than the superior Eicon Diva Server 4BRI-8M.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Mon, 2003-09-15 um 15.02 schrieb Matthew Enger:
> Hello,
> 
> Not sure if it will do what you want, but have you looked at the AVM C2
> or C4 ISDN cards (www.avm.de)? They work under capi on linux from what I
> have read and that is what my work is about to use as soon as we can
> source them in Australia (supplier here has not responded so far).
> 
> Regards,
>   Matthew Enger
>   [EMAIL PROTECTED]
> 
> On Mon, 2003-09-15 at 19:42, Jean-Marc V. Liotier wrote:
> > Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking
> > for a CAPI compatible active BRI adapter capable of NT mode. I have
> > found that the 'Eicon Diva Server 4BRI' fulfills theses requirements.
> > However, at EUR 1387 for four BRI interfaces, it does not come cheap.
> > Are there other users of similar hardware that can point me to
> > alternatives ?
> > 
> > 
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Re: [Asterisk-Users] Anyone using National ISDN (NI-1) BRI under Linux?

[Klaus-Peter Junghanns]

Hi Steve,

the Eicon Diva Server BRI cards support NI-1 with capi. Some people in
.us are using them. Unfortunately AVM only put EuroISDN into their
binary kernel module. :(

Are you sure that incoming callerid is included in your "FastTrak ISDN"?
i could imagine that SBC will rip you off extra for it! ;)

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mon, 2003-09-15 um 23.10 schrieb Steve Haehnichen:
> I have a North American BRI configured as National ISDN (NI-1) on an
> SBC (Nortel?) switch.  SBC calls this "FastTrak ISDN", and it's
> surprisingly inexpensive -- less than two POTS lines.
> 
> I've been trying to find an inexpensive PCI interface solution to
> connect this to Asterisk as voice lines.  I prefer ISDN BRI over POTS
> lines because of the improved signalling, fast dialing, clean audio,
> and so on.
> 
> I've tried three different Passive ISDN boards supported by the HiSax
> drivers, with various degrees of failure.  None have been able to
> complete a call.  (Linux kernel 2.4.22, with AVM Fritz, DynaFax, and
> HFC cards)
> 
> According to the HiSax documentation, NI-1 is a supported protocol.
> 
> From what I can find on CAPI, it supports only EuroISDN:
>   http://www.junghanns.net/asterisk/page15.html
> Is this the case for all Linux CAPI drivers?  Bummer.
> 
> Has anyone successfully used Asterisk with a HiSax passive ISDN card
> on a National-1 (or NI-2) dialing plan?  It would really mean a whole
> lot just to hear that someone is doing it with success. :)
> 
> 
> How does one specify a SPID in Asterisk?  I've done an archive search
> and came up blank for "spid".  BRI (2B+D) provisioning on SBC is done
> with two Directory Numbers and two SPIDs, like this:
> 
> DN1: 794-7601
> DN2: 794-7602
> 
> SPID1: 858-794-7601-0101
> SPID2: 858-794-7602-0101
> 
> (SBC said the -0101 is used on all NI-1 SPIDs)
> 
> I'm not sure how to inform Asterisk/HiSax of the SPID part, and I get
> a "SPID not supplied in EAZMSN" from the ISDN driver.
> 
> I've tried various permutations of AT&E7947601:85879476010101 manually
> into the /dev/ttyI0 interface, with no luck.
> 
> Last, according to the kernel messages, there never seems to be an OAD
> (Caller-ID) on incoming calls.  I've used an ISDN protocol analyzer to
> show that the incoming setup does contain the calling number.  Again,
> I'm just curious to compare notes with someone else doing this.
> 
> Thanks!
> 
> -Steve
> 
> My modem.conf:
> 
> [interfaces]
> context=isdn-in
> driver=i4l
> language=en
> type=autodetect
> stripmsd=0
> dialtype=tone
> mode=immediate
> 
> group=1
> msn=7947601
> incomingmsn=*
> context=isdn-in
> device => /dev/ttyI0
> device => /dev/ttyI1
> 
> ---
> 
> My Kernel log:
> 
> ISDN subsystem Rev: 1.1.4.1/1.1.4.1/1.1.4.1/none/1.1.4.1/1.1.4.1 loaded
> HiSax: Linux Driver for passive ISDN cards
> HiSax: Version 3.5 (module)
> HiSax: Layer1 Revision 1.1.4.1
> HiSax: Layer2 Revision 1.1.4.1
> HiSax: TeiMgr Revision 1.1.4.1
> HiSax: Layer3 Revision 1.1.4.1
> HiSax: LinkLayer Revision 1.1.4.1
> HiSax: Total 1 card defined
> HiSax: Card 1 Protocol NI1 Id=HFC (0)
> HiSax: HFC-PCI driver Rev. 1.1.4.1
> HiSax: HFC-PCI card manufacturer: CCD/Billion/Asuscom card name: 2BD0
> HFC-PCI: defined at mem 0xe095dc00 fifo 0xd9e88000(0x19e88000) IRQ 5 HZ 100
> HFC_PCI: resetting card
> HFC 2BDS0 PCI: IRQ 5 count 0
> HFC 2BDS0 PCI: IRQ 5 count 34
> HiSax: National ISDN-1 Rev. 1.1.4.1
> HiSax: National ISDN-1 Rev. 1.1.4.1
> HiSax: 2 channels added
> HiSax: MAX_WAITING_CALLS added
> ...
> isdn_net: call from 0 -> 0 7947601 ignored
> isdn_tty: Incoming call without OAD, assuming '0'
> isdn_tty: call from 0, -> RING on ttyI1
> SPID not supplied in EAZMSN 
> isdn: HFC,ch0 cause: 001B
> isdn: HFC,ch1 cause: 6F15
> isdn_net: Incoming call without OAD, assuming '0'
> isdn_net: call from 0 -> 0 7947601 ignored
> isdn_tty: Incoming call without OAD, assuming '0'
> isdn_tty: call from 0, -> RING on ttyI0
> isdn_net: Incoming call without OAD, assuming '0'
> isdn_net: call from 0 -> 0 7947601 ignored
> isdn_tty: Incoming call without OAD, assuming '0'
> isdn_tty: call from 0, -> RING on ttyI1
> isdn: HFC,ch0 cause: 001B
> isdn_net: Incoming call without OAD, assuming '0'
> isdn_net: call from 0 -> 0 7947601 ignored
> isdn_tty: Incoming call without OAD, assuming '0'
> isdn_tty: call from 0, -> RING on ttyI0
> SPID not supplied in EAZMSN 
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Re: [Asterisk-Users] Re: ISDN BRI active adapters with NT mode - any alternatives ?

[Klaus-Peter Junghanns]

Am Die, 2003-09-16 um 18.05 schrieb Louis-David Mitterrand:
> I am using the Diva 4BRI daily with our * and indeed it does support NT
> mode on a port by port basis: when you configure the card initially you
> are specifically asked whether you want ports in TE or NT mode. And this
> is with the open-source Melware drivers from http://mmm.melware.de.
> 
> Now I have no idea if * supports plugging ISDN phones in the Diva. AFAIK
> it's not supported by chan_capi, but that may change.
> 

Yes, that may change. I will check with Eicon headquarters what the NT
mode support in the BRI cards is about. And if somebody out there feels
like sponsoring an Eicon BRI, so i can add support for it to chan_capi
i wouldnt mind taking it ;-) (a place in the capi hall of fame will be
yours) ;-)


regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Re: [Asterisk-Users] Need help with H.323

[Klaus-Peter Junghanns]

Am Don, 2003-09-18 um 16.36 schrieb Roy Sigurd Karlsbakk:
> hi all

hi roy,

> 
> I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with
> the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h
> software can only be configured to point to a gatekeeper. I know I don't
> need to do this, but it's a test before I setup my Symbol Netvision
> phones (I don't have an access point for them now). First, this is what
> I have, and what I've done:
> 
> - Platform is Debian Woody/Stable
> - Asterisk is fresh from CVS, and so is chan_h323
> - chan_h323 did compile, but also died from a rather nasty SIGSEGV
>   (reported earlier) with Open H.323 v1.11.7 and PWLib v1.4.11. I
>   checked them out from CVS instead after being adviced so by diana
>   (on irc), and it now loads into memory without any fuzz.
> - Several people have told me chan_capi can work as a gatekeeper, so
>   there should be no use for gnugk or any others. I have yet to find
>   where this is hidden. FWICS, this is all commented out (ast_h323.cpp
>   line 722). Is this right or have I overseen anything?

i can confirm that chan_capi will not work as a gatekeeper under any
circumstances! somebody is obviously trying to confuse you.;-)

> - If this cannot be done alone with chan_h323, I guess I need to use
>   gnugk or something. Can someone help me out how these two should be
>   configured to work together? Also - is it possible to run gnugk on
>   the same host as asterisk?
> 
> Thanks for all help
> 
> Best regards
> 
> Roy Sigurd Karlsbakk <[EMAIL PROTECTED]>
> 

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
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[Asterisk-Users] chan_capi 0.2.5b released

[Klaus-Peter Junghanns]

Hi capi users,

chan_capi 0.2.5b brings some new options to simplify your capi.conf.

msn= is now a comma separated list of allowed outgoing MSNs

controller= takes a comma separated list too (no need to copy your
capi interfaces for multiple controllers anymore)

incomingmsn=* catches every incoming MSN that pops up on your line.

get it at http://www.junghanns.net/asterisk/

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
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Re: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0

[Klaus-Peter Junghanns]

Hi Jim,

get chan_capi from www.junghanns.net/asterisk/

install the capi drivers for your card from ftp.avm.de/cardware

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mit, 2003-09-24 um 21.17 schrieb Jim Paraschou:
> Hi,
> 
>   I use an AVM FRITZ PCI 2.0 to dial out but although
> it works OK and places the call there is no ring or
> busy tone.
>   Has someone figured out this problem?
>   Thanks
> 
> 
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Re: [Asterisk-Users] Chan_capi accountcode.. (repost)

[Klaus-Peter Junghanns]

Hi,

update to chan_capi 0.2.5c and add accountcode=bla to your
capi devices.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mit, 2003-09-24 um 22.53 schrieb WipeOut .:
> Hi,
> 
> All my inbound calls have a blank account code in the CDR..
> 
> Where or what is the correct way to set the "accountcode=" setting when using 
> chan_capi channels?
> 
> Do I do it in capi.conf or extensions.conf with a setvar? or some other way..
> 
> Thanks..
> 
> -- 
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Re: [Asterisk-Users] Ascom Ascotel 2050 & Fritz PCI Card (Capi)

[Klaus-Peter Junghanns]

Hi Dave,

try :

Dial(CAPI/${CALLERIDNUM}:${EXTEN},10)

(make sure you have a msn= line in capi.conf that allows
this for ${CALLERIDNUM})

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Fre, 2003-10-03 um 18.58 schrieb Dave Sykes:
> Hello,
> 
> We have been trying to add asterisk to our Ascom Ascotel 2050 PBX. We have
> a AVM Fritz!PCI Card connected to an S0 bus extension from the PBX. The
> fritz card is configured to use chan_capi, and we can make calls SIP->SIP
> SIP->PBX extension PBX extension->SIP all successfully, we have assigned
> more than one PBX extension number to the S0 port in the Ascom PBX (it has
> 8 positions) calls from the PBX to the SIP phones work fine and are
> directed correctly. However when we dial from a SIP phone to an Ascom
> extension it always comes up on the Ascom handset with the number of the
> first position in the S0 setup.
> 
> For Example
> 
> S0 setup in Ascom
> position 1: 616
> position 2: 617
> position 3: 618
> 
> in extensions.conf
> 
> exten => 1,1,Dial(Sip/X616,10)
> exten => 2,1,Dial(Sip/X617,10)
> exten => 3,1,Dial(Sip/X618,10)
> 
> calls from an Ascom extension to 616,617,618 get directed correctly to the
> Sip phone X616,X617,X618 respectively
> 
> for outgoing calls we have
> 
> exten => _XXX,1,Dial(CAPI/@01234567${CALLERIDNUM}:${EXTEN},10)
> 
> and dialling for instance 765 from a Sip phone correctly routes the call to
> the Ascom and that then connects through to extension 765, however no
> matter which Sip phone dials it always displays 616 on the Ascom phone as
> the Caller ID. I can see from the * console that it has picked up the
> caller id correctly
> for example to dial 765 from Sip/X617
> -- Executing Dial("SIP/X617-5ee5", "CAPI/@01234567617:765|10") in new stack
> 
> So the question is: is it possible to tell the Ascom exchange that the call
> is coming from extension 617, as I guess it does not use the caller id as
> that seems to be set up correctly?
> 
> Thanks
> 
> Dave Sykes
> Head of Research
> +44 (0)1252 740721
> 
> 
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Re: [Asterisk-Users] chan_capi and latest Debian package

[Klaus-Peter Junghanns]

Hi capi users :-)

you might also want to try chan_capi 0.3.0 which is already
in the downloads directory but not linked on the page.
The option echosquelch=1 now finally works.

regards

kapejod
-- 
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CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
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Am Mit, 2003-10-08 um 16.18 schrieb Peer Oliver schmidt:
> Peer Oliver schmidt wrote:
> > After
> > 
> > apt-get update && apt-get upgrade -y
> > wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz
> > tar xfvz chan_capi.02.5c.tar.gz
> > cd chan_capi-0.2.5c
> > make && make install
> > shutdown -r now
> > 
> > asterisk seg faults upon calling in via ISDN.
> 
> After another
> apt-get install asterisk-dev
> another make cycle for chan_capi
> apt-get install asterisk
> 
> everything is working again.
> 
> Sorry for the traffic.
> 
> rgds
> pos
> 
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Re: [Asterisk-Users] Sasquatch, the Loch Ness Monster, UFOs and...

[Klaus-Peter Junghanns]

dont forget "generic voice modem" support. or even better
"chan_modoss" or "chan_modalsa" the combination of an
external modem (for signalling) and a sound card :)


Am Don, 2003-10-09 um 11.54 schrieb Michael Bielicki:
> wan't to add DS3 and SS7 to that ?
> also licensed g723.1 and working g729
> softfax and softmodem
> 
> that's what comes to my mind on the spot ...
> 
> On Thursday 09 October 2003 9:51 am, John Todd wrote:
> > Mythical Asterisk Creatures, oft-discussed, rarely seen:
> >
> > 1) An "advanced" graphical user interface
> >
> > 2) An IAX2 hardware device
> >
> > 3) A Radius CDR report module
> >
> > 4) A live-method, robust SQL-based dialplan
> >
> > 5) LDAP/SQL/Radius authentication for SIP phones
> >
> > 6) Robust R2 signalling support
> >
> > 7) Multilingual language recordings of all existing * .gsm files
> >
> > 8) Free exchange of PSTN gateways in a centralized routing arbiter model
> >
> > 9) Speech recognition support
> >
> >
> > Care to add your own unicorns to the list?  I make no judgement nor
> > do I cast aspersions on any of these items, but I seem to recall
> > seeing comments about "I'm working on..." or "It would be really
> > great if..." on all of these without seeing real evidence on any of
> > them other than talk.  The only well-remembered myth I can say for
> > certain that has been dispelled is the SCCP channel driver, and that
> > has been moved out of "Loch Ness" status to "peer-reviewed" status.
> >
> > JT
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> -- 
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> Managing Director
> TAAN Consultants Ltd
> http://www.global-gateway.net/
> 
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> 
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> 
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Re: [Asterisk-Users] chan_capi and latest Debian package

[Klaus-Peter Junghanns]

Hi,

echosquelch=1 enables Petr Michalek's "echo canceler", which
compares RX and TX volumes and mutes the RX in an echo condition.

regards

kapejod
-- 
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CEO,CTO
Junghanns.NET GmbH
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fon:+49 30 79705392
fax:+49 30 79705391
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Am Don, 2003-10-09 um 09.09 schrieb Florian Overkamp:
> At 18:45 8-10-2003 +0200, you wrote:
> >Hi capi users :-)
> >
> >you might also want to try chan_capi 0.3.0 which is already
> >in the downloads directory but not linked on the page.
> >The option echosquelch=1 now finally works.
> 
> Yeah, I found 0.3.0 recently and installed it, seems to be working fine. 
> What is echosquelch supposed to do ?
> 
> Florian
> 
> 
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Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)

[Klaus-Peter Junghanns]

Hi Marcel,

IE 76 is COLP (Connected Line ID Presentation).
Your telco is so kind to tell you to which number your calls has
been connected. Noting to worry about...

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mon, 2003-10-13 um 16.56 schrieb Marcel Prisi:
> Here is an example call (works) :
> 
>  -- Executing Dial("SIP/25-e804", "Zap/g1/0707038340") in new stack
>  -- Called g1/0707038340
>  -- Zap/1-1 is ringing
> !! Unknown IE 76 (Unknown Information Element)
>  -- Zap/1-1 answered SIP/25-e804
> 
> What does that "!! Unknown IE 76 (Unknown Information Element)" mean ??
> 
> Thanks
> 
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Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)

[Klaus-Peter Junghanns]

Hi Martin,

it's not implemented in libpri but very well standarized (ETS 300 097).

regards,
kapejod

-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mon, 2003-10-13 um 17.24 schrieb Martin Pycko:
> It means that this IE is not implemented in the libpri or is not very
> standarized.
> 
> regards
> Martin
> 
> On Mon, 13 Oct 2003, Marcel Prisi wrote:
> 
> > Here is an example call (works) :
> >
> >  -- Executing Dial("SIP/25-e804", "Zap/g1/0707038340") in new stack
> >  -- Called g1/0707038340
> >  -- Zap/1-1 is ringing
> > !! Unknown IE 76 (Unknown Information Element)
> >  -- Zap/1-1 answered SIP/25-e804
> >
> > What does that "!! Unknown IE 76 (Unknown Information Element)" mean ??
> >
> > Thanks
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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Re: [Asterisk-Users] Strange message !! Unknown IE 76 (Unknown Information Element)

[Klaus-Peter Junghanns]

Hi Martin,

libpri misses all the fun stuff :-(
hold, retrieve, suspend, ect, cd, conf, 3pty ..

but i am going to change that :-)

regards
kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mon, 2003-10-13 um 18.11 schrieb Martin Pycko:
> My fault then :)
> I was thinking only in terms of Q931 spec ...
> 
> Martin
> 
> On 13 Oct 2003, Klaus-Peter Junghanns wrote:
> 
> > Hi Martin,
> >
> > it's not implemented in libpri but very well standarized (ETS 300 097).
> >
> > regards,
> > kapejod
> >
> > --
> > Klaus-Peter Junghanns
> >
> > CEO,CTO
> > Junghanns.NET GmbH
> > Breite Strasse 13 - 12167 Berlin - Germany
> > fon:+49 30 79705392
> > fax:+49 30 79705391
> > iaxtel: 1-700-157-8753
> > email:  [EMAIL PROTECTED]
> > http://www.junghanns.net/asterisk
> >
> > Am Mon, 2003-10-13 um 17.24 schrieb Martin Pycko:
> > > It means that this IE is not implemented in the libpri or is not very
> > > standarized.
> > >
> > > regards
> > > Martin
> > >
> > > On Mon, 13 Oct 2003, Marcel Prisi wrote:
> > >
> > > > Here is an example call (works) :
> > > >
> > > >  -- Executing Dial("SIP/25-e804", "Zap/g1/0707038340") in new stack
> > > >  -- Called g1/0707038340
> > > >  -- Zap/1-1 is ringing
> > > > !! Unknown IE 76 (Unknown Information Element)
> > > >  -- Zap/1-1 answered SIP/25-e804
> > > >
> > > > What does that "!! Unknown IE 76 (Unknown Information Element)" mean ??
> > > >
> > > > Thanks
> > > >
> > > > ___
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> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > > ___
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> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
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> >
> 
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Re: [Asterisk-Users] ISDN PBX + IVR + Voicemail Configuration - Sanity Check ...

[Klaus-Peter Junghanns]

Hi Hans,

Am Don, 2003-11-06 um 15.58 schrieb Vledder, Hans:
> P.S. He who comes up with clean internal ISDN bus (point to multi-point)
> support for Asterisk, based on CologneChip based equipment receives an 18"
> large Dutch cheese in the mail, right after I've wiped away my tears of
> happiness !
> 

I am currently working on a zaptel driver for the hfc-s pci a based ISDN
cards and the modifications to make libpri work with BRIs. And of course
NT mode will be supported. We will also have a 4 port BRI card available
in mid/late november that works in TE and NT mode (onboard termination).
Check out www.junghanns.net/asterisk/ during the next week.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

P.S. i have no idea where i should put the 18" cheese ;-) if it was 19"
i could bring it into the colo...
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Re: [Asterisk-Users] Eicon Diva Server 4BRI

[Klaus-Peter Junghanns]

Hi Lars,

the Eicon Diva Server 4BRI works with the GPL capi drivers from
mmm.melware.de and chan_capi, echo cancelation is supported and
works like a charm. stay away from i4l ;-)

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Son, 2003-11-09 um 02.54 schrieb Lars Boegild Thomsen:
> Hi Everybody,
> 
> Has anybody tried the above (or indeed any other 4XBRI cards) successfully
> with Asterisk.  As far as I can see the above mentioned card is an active
> ISDN card but supported by it's own I4L driver.  This leads to interesting
> questions particularly regarding echo cancellations (which usually doesn't
> work on the cheap passive cards with one exception as far as I can see).
> 
> Regards,
> 
> Lars...
> 
> --
> Lars Boegild Thomsen
> Technical Director
> JustIT Sdn. Bhd.
> Cell Phone (MY): +60 (16) 323 1999
> ICQ: 6478559
> Yahoo Chat: [EMAIL PROTECTED]
> MSN Chat: [EMAIL PROTECTED]
> http://www.justit.ws
> 
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Re: [Asterisk-Users] unable to find path

[Klaus-Peter Junghanns]

Hi Kevin,

you are probably using an IP phone which is configured to use
g729 as the codec. Configure your phone to use ulaw (a.k.a. g711
or PCMU). If you want to make use of g729 to save bandwidth when
connecting over a WAN you can purchase a g729 license at digium.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

 
Am Son, 2003-11-09 um 11.21 schrieb Kevin Bockman:
> Hi. I just tried updating asterisk and I guess I broke something.  Here's the log:
> 
> Unable to find a path from G729A to SLINR
> Unable to find a path from ULAW to G729A
> 
> Any ideas on what I should try?  I tried nuking all the zaptel stuff in the system 
> and the source and started over agian.  Also nuked the asterisk config files
> 
> I saw this asked once before but there was no reply :-/
> 
> Thanks,
> 
> Kevin
> 
> _
> Are you a Techie? Get Your Free Tech Email Address Now! Visit 
> http://www.TechEmail.com
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Re: [Asterisk-Users] Fax over SIP alaw/ulaw

[Klaus-Peter Junghanns]

Am Sam, 2003-11-15 um 11.22 schrieb Florian Overkamp:
> At 20:34 14-11-2003 -0600, you wrote:
> >Should I expect a standard fax machine connected to an ata-188 connected
> >to an asterisk server, connected to a pri fed from a cisco 7206vxr to work
> >correctly? It needs to have a standard fax machine, receiving and emailing
> >it won't be acceptable.
> 
> I have gotten this to work, but it seems to be very dependant on your 
> infrastructure. Fax over alaw or ulaw clear channels will work as long as 
> the latency of the network is within acceptable range. Give it a try :-)
> 
> Florian

On a LAN this should work without problems. I am even faxing over the
wild internet with ping times between 100 and 120 ms. Both locations
have an AVM Fritz card connected to an internal S0 bus of an isdn pbx
and run chan_capi.

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
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Re: [Asterisk-Users] FYI: Simple Small Asterisk install..

[Klaus-Peter Junghanns]

Hi,

if anyone is looking for a really small * installation take a SuSE
rescue system, add sshd, zaptel and *. Gzip it and the image is only
28MB which fits into a 64MB ramdisk and leaves some MBs for temporary
logs, CDRs and voicemail.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

Am Mit, 2003-11-19 um 16.41 schrieb WipeOut:
> Hi,
> 
> If anyone is looking for a small Asterisk installation I have managed to 
> get it down to 296MB (If you remove the kernel source code.. could 
> probably be made smaller if some of the devel packages and asterisk 
> source is removed as well.)
> 
> To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a 
> minimum install with ssh support(92MB) and then added the required 
> packages individually..
> 
> Everything has compiled and Asterisk loads but I haven't tried any 
> Zaptel drivers or hardware so can't comment on that..
> 
> Anyway if anyone needs a small secure install there it is..
> 
> Later..
> 
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Re: [Asterisk-Users] FYI: Simple Small Asterisk install..

[Klaus-Peter Junghanns]

Hi,

you can boot with your favourite bootloader. I boot with pxe over the
network or with syslinux from a 32mb compact flash card usually.
Make sure you add "root=/dev/ram initrd=image.gz" to your kernel 
append line.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

> I know this may seem like a dumb question, how do you run the system 
> from the gzipped file? got any links to info?
> 
> Guess I need to get to know liinux better.. :)
> 
> Later..
> 
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Re: [Asterisk-Users] SIP (peer to peer?)

[Klaus-Peter Junghanns]

> 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 if the both ends have incompatible codec settings and Asterisk is able to 
> translate.
> 
or if the canreinvite= parsing in chan_sip.c is broken...

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RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[Klaus-Peter Junghanns]

Hi,

faxing works great with IAX/IAX2. Even over a 100ms ADSL link.
It does not really depend on the protocol, only on the codec.

regards

kapejod

Am Fr, 2003-12-12 um 14.42 schrieb Senad Jordanovic:
> Dan wrote:
> > Hi,
> > 
> > It would be great if the IAX protocol will be able to tranfer fax
> > data (even converted in another format) between Asterisk boxes, using
> > low bandwidth codecs like GSM. I know that this is possible only with
> > the G.711 now (passing faxes using the audio stream), but maybe
> > in the future...some native support will permit this.
> > 
> > Just a thought,
> > Dan
> > 
> > ___
> > Asterisk-Users mailing list
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> My understanding is that fax calls are ONLY possible with G711/SIP not
> IAX.
> Anyone?
> 
> Ta
> SJ
> 
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Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)

[Klaus-Peter Junghanns]

wasim: yes i am! actually i always was ;-)

ok, here is my setup

Fax machine -> ISDN pbx -> (chan_capi) -> Asterisk 1 -> LAN(chan_iax2)
-> Asterisk 2 -> Internet 100ms (chan_iax2) -> Asterisk 3 -> (chan_capi)
-> ISDN pbx -> Fax machine

i use plain iax2, no trunking.

regards

kapejod

Am Fr, 2003-12-12 um 15.15 schrieb Andrew Thompson:
> - Original Message -----
> From: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, December 12, 2003 9:02 AM
> Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
> 
> 
> > Hi,
> >
> > faxing works great with IAX/IAX2. Even over a 100ms ADSL link.
> > It does not really depend on the protocol, only on the codec.
> >
> > regards
> >
> > kapejod
> >
> 
> Can you give us some detail about your config to do this?
> 
> Either Dan's not tried it (maybe he didn't think it would work), or couldn't
> make it work.
> 
> -
> Andrew Thompson http://aktzero.com/
> Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
> restful it is to watch the cursor blink. Close your eyes. The opinions
> stated above are yours. You cannot imagine why you ever felt otherwise.
> 
> 
> 
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Re: [Asterisk-Users] CLIP in Germany

[Klaus-Peter Junghanns]

Hi Johannes,

i havent tried it myself, but in .de phones use the same FSK-style
callerid like in the US. I have seen .de CLIP phones working with
an ATA186, so the digium FXS cards should do that too.

best regards

kapejod

Am Fr, 2003-12-12 um 12.51 schrieb Johannes von Drachenfels:
> Hi,
> 
> does anybody has experience with using CLIP for FXS-phones with E100P and
> TDM400 installed ?! Any little help would be great !
> 
> Thanks,
> 
> Johannes
> 
> **
> Johannes von Drachenfels  Telefon:+49 7231 922380 0
> Drachenfels GmbH  Fax:+49 7231 922380 3
> Bleichstrasse 56  Mobile: +49 171 6710815
> 75173 Pforzheim   Mail:   mailto:[EMAIL PROTECTED]
> Germany   WWW:http://www.drachenfels.de
> **
> 
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Re: [Asterisk-Users] chan_capi 0.3 and capiinit questions

[Klaus-Peter Junghanns]

Hi,

1) you are doing "capiinit stop/start" while * is running?
do you think that unloading a device driver while * is still
running is a good idea? ;-)

2) there is no (clean) way to determine if another application
is using a B channel. However you will get a "circuit-busy" 
reason back from capi, if you hack that into chan_capi / * you
might be able to recognize that the outgoing call failed because
all B channels are in use.

regards 

kapejod

Am Fr, 2003-12-12 um 17.41 schrieb Philipp von Klitzing:
> Hi there,
> 
> two questions concerning ISDN BRI, Fritz! passive and chan_capi:
> 
> 1. I noticed that I run into trouble if I do a "capiinit stop" and 
> "start" while Asterisk is running. Is that normal, or do I need to twist 
> my configuration somehow?
> Background: ISDN callers normally get a voice menu, however my box would 
> accept the call, show on the CLI that it was playing the voice menu, but 
> the caller would hear nothing. Restarting * did not help, instead a stop 
> now plus capiinit restart solved this.
> 
> 2. If I want to use one of the two B-channels as Internet dial-up backup 
> in case the cable link is down, will I have to notify * that I took away 
> one of the two channels? I noticed that "capi info" in the CLI will show 
> "2 channels free" even if another application outside * uses on of the 
> two channels...
> 
> Cheers, Philipp
> 
> 
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Re: [Asterisk-Users] E400 or TE410 (digium) vs PRI 30M (Eicon)

[Klaus-Peter Junghanns]

Hi,

the Eicons work fine with chan_capi, also the hardware echo
cancelation works fine.

regards

kapejod

Am Mo, 2003-12-15 um 17.38 schrieb Steven Critchfield:
> On Mon, 2003-12-15 at 09:58, Daniel ANDRE wrote:
> > Hello,
> > 
> > I would like to have some comparison between E1 cards from Digium and 
> > those from Eicon for a VOIP - ISDN Gateway.
> > 
> > How does they compare on the echo cancel point of view?
> > Is the echocancellation code for E400 good enough for production 
> > environment?
> 
> The code for the Digium E1 is the same as used on any other Digium
> interfaces. It is in software and shared across all Zaptel hardware.
> There shouldn't be an echo at the E1 interface. The echo will be on
> either an analog end or in speaker bleed over to mic. 
> 
> Something you will also need to consider, what software will you be
> using? Digium cards work with asterisk, and I doubt that the Eicon E1
> cards do yet unless they use the capi driver. 
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[Asterisk-Users] Zaptel BRI experimental drivers released.

[Klaus-Peter Junghanns]

Hi isdn folks,

i just released some (still) experimental drivers for Zaptel
BRI isdn support. Currently it includes kernel modules for our
quadBRI PCI ISDN card. A driver for the HFC-S PCI A chipset based
el-cheapo isdn cards will follow soon.
We now also have the quadBRI cards in stock.

Find the drivers at:
http://www.junghanns.net/asterisk/page17.html

regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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RE: [Asterisk-Users] Free Software/Open Source-Telephony-Summit 2004

[Klaus-Peter Junghanns]

Hi,

since there will be people from around the globe it will all
be done in English.

regards

kapejod

> Hi Philipp-
>
> Just out of curiosity, are these types of workshops generally conducted
> in German, or in English?
>
> Cheers
> Scott
>
> London
>
> Scott M. Stingel
> Emerging Voice Technology Inc.
>
> Email:  [EMAIL PROTECTED] 
> URL:www.evtmedia.com 
>
>
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> Philipp von Klitzing
>> Sent: Tuesday, December 16, 2003 3:35 PM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] Free Software/Open
>> Source-Telephony-Summit 2004
>>
>>
>> Hi,
>>
>> I just came across this annoncement, which is particularly
>> interesting as
>> it is only 25 min away from my place... :-) Anyway, I guess
>> the core of
>> this is targeted at developers mainly.
>>
>> Cheers, Philipp
>>
>>
>> Free Software/Open Source-Telephony-Summit 2004
>>
> http://www.guug.de/veranstaltungen/telephony-summit-2004/
> http://www.heise.de/newsticker/data/avr-16.12.03-000/
>
> We are happy to announce that the first summit on Free Software/Open
> Source-telephony solutions is going to take place from January 16th till
>  20th in Geilenkirchen, Germany.
>
> The event will be divided into three parts:
>
> a developer workshop from January 16th to January 18th
> a conference day (January 19th)
> a tutorial day (January 20th)
> There will be an exhibition during the conference and the tutorial day.
>
> The developer workshop is free of charge and only-open for active
> developers in Free Software/Open Source telephony projects. If you are
> interested in participating, please contact Martin Schulte telephony-
> [EMAIL PROTECTED]
>
> During the one-day conference, the participating projects will give an
> overview about their current status and their future goals.
>
> In the morning of the tutorial day, people interested in using a
> particular software will get an in depth-introduction to installation,
> configuration and usage by their developers. In the afternoon, there's a
>  big "How everything works together"-tutorial.
>
> Online-Registration is opened!
>
>
> ___
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>
>
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Re: AW: [Asterisk-Users] Capi Dial & outgoing msn?

[Klaus-Peter Junghanns]

> Hi,
>
> On Tue, 2003-12-23 at 19:01, Patrick wrote:
>> On Tue, 2003-12-23 at 17:13, [EMAIL PROTECTED] wrote:
>> > try it without prefix (else dtag uses first msn) -
>> > so if your city code is 07032 and phone no (msn) 41432
>> > -> exten => _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
>
>> Thanks for the pointer Thomas. I removed the areacode from msn= in
>> capi.conf and from the dial statement. Tried again and till no CLID.
>> Stumped at this point. Perhaps my telco doesn't allow setting outgoing
>> msn numbers.
>
> In the Netherlands you would send the areacode with the leading zero
> removed to make this work, so Dial(Capi/@703241432:${EXTEN}|30|r)
> following the earlier example. Different countries have different ways
> to do it. Checkout the FAQ on www.isdn4linux.de, it has information for
> many different countries.
>

If you use "@" before the outgoing MSN you will enable CLIR. So the other
party will not see your caller id.

regards
kapejod



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[Asterisk-Users] FS/OS Telephony Summit 2004

[Klaus-Peter Junghanns]

Hello * world,

i will be attending the FS/OS Telephony Summit 2004 in Geilenkirchen
from the 16th til 20th january. Together with Christian Richter i will
be speaking about * on monday. And we will give an * tutorial on
tuesday. I will be presenting some ISDN stuff there, including the
quadBRI cards.
If you will be there too and want to meet, just let me know. :)

Details on the summit can be found at:
http://www.guug.de/veranstaltungen/telephony-summit-2004/

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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Re: [Asterisk-Users] newbie ISDN question

[Klaus-Peter Junghanns]

Hi Thorsten,

the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect
phones to that card.
The quadBRI card has 4 BRI ports that can individually be configured
for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones).
Please find the details at:

http://www.junghanns.net/asterisk/page17.html

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/

> hi everybody, sorry for posting such a stupid question ;)
>
> i've managed to run asterisk* with my AVM fritz2.0 card and a some
> VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied
> me ;)))
>
> now i want to run asterisk* istead of our old PBX. but it would be great
> to connect some phones directly to my box. how does a E100P from digium
> work. can i connect it to my ISDN-line and my internal phones (ISDN)?
>
> it would look like this:
>
> [PHONE2]
>  /
> [PC]-[E100P]  - [PHONE1]
>  \
>  [ISDN-LINE]
>
> thank you for your help!!!
> thorsten
>
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Re: [Asterisk-Users] newbie ISDN question

[Klaus-Peter Junghanns]

The quadBRI card is EUR 600, excluding VAT.

best regards

kapejod

> Hello kapejod,
>
>> The quadBRI card has 4 BRI ports that can individually be configured
>> for TE mode (to connect ISDN lines) or NT mode (to connect ISDN
>> phones). Please find the details at:
>>
>> http://www.junghanns.net/asterisk/page17.html
>
> when are you going to release some pricing on the card? It just says
> "But me!", but does not show you how... :)
>
> rgds
> pos
>
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Re: [Asterisk-Users] newbie ISDN question

[Klaus-Peter Junghanns]

Thorsten,

theoretically you can connect 8 phones per port, but only 2 can
be used at the same time. We advise to use 2 per port and in
some scenarios 3 might be an option. So you can connect 8 ISDN
phones to the quadBRI card.
The drivers are still released as "experimental" and have some
bugs. We are planning to be stable in about 2 weeks.

The cards are in stock, so delivery will be fast. We ship with
worldwide with UPS.

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/

> hi klaus-peter,
>
> thank you for your replay. btw: i am using you chan_capi already ;)) it
> works great!!!
> how many internel phones could be connected to this card?
> how stable is the driver (can i use it for a production-system)?
>
> sorry for all that stupid questions - i know linux and ip and
> pc-hardware but telephone-technics are all new for me.
>
> how long would delivery of that card take?
>
> thanks (oder besser gesagt: VIELEN DANK ;)  )
> thorsten
>
> - Original Message -
> From: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, January 14, 2004 11:54 AM
> Subject: Re: [Asterisk-Users] newbie ISDN question
>
>
>> Hi Thorsten,
>>
>> the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect
>> phones to that card.
>> The quadBRI card has 4 BRI ports that can individually be configured
>> for TE mode (to connect ISDN lines) or NT mode (to connect ISDN
>> phones). Please find the details at:
>>
>> http://www.junghanns.net/asterisk/page17.html
>>
>> best regards
>>
>> kapejod
>> --
>> Klaus-Peter Junghanns
>>
>> CEO, CTO
>> Junghanns.NET GmbH
>> Breite Straße 13 - 12167 Berlin - Germany
>> fon: (de) +49 30 79705390
>> fon: (uk) +44 870 1244692
>> fax: (de) +49 30 79705391
>> iaxtel: 1-700-157-8753
>> http://www.Junghanns.NET/asterisk/
>>
>> > hi everybody, sorry for posting such a stupid question ;)
>> >
>> > i've managed to run asterisk* with my AVM fritz2.0 card and a some
>> VOIP-softphones (SIP, H323). the functions of asterisk* really
>> satisfied me ;)))
>> >
>> > now i want to run asterisk* istead of our old PBX. but it would be
>> great to connect some phones directly to my box. how does a E100P
>> from digium work. can i connect it to my ISDN-line and my internal
>> phones (ISDN)?
>> >
>> > it would look like this:
>> >
>> > [PHONE2]
>> >  /
>> > [PC]-[E100P]  - [PHONE1]
>> >  \
>> >  [ISDN-LINE]
>> >
>> > thank you for your help!!!
>> > thorsten
>> >
>> > ___
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>> >http://lists.digium.com/mailman/listinfo/asterisk-users
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>>
>>
>>
>>
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>
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Re: [Asterisk-Users] Re: newbie ISDN question

[Klaus-Peter Junghanns]

Hi,

yes, for the home user it's still too expensive. Although it's
really cheap if you compare it to other 4 BRI cards on the market.

Currently i am polishing the driver for the hfc-s pci a chipset,
which i used in numerous el-cheapo ISDN cards (street price around
30 EUR). This will bring zaptel BRI (and even NT mode) to the
home user. :)

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


> Hi,
>
> On Wed, 14 Jan 2004 at 12:15, Klaus-Peter Junghanns wrote:
>
>> The quadBRI card is EUR 600, excluding VAT.
>
> this looks like a great piece of hardware, but I think it's too
> expensive for home users like me who wouldn't really need more than one
> or two BRI ports.
>
> So do you have any plans for a singleBRI or doubleBRI version of this
> card, or maybe even a variant that comes with a single port
> preinstalled and three more ports can be added as needed via
> daughterboards like on the TDM400P?
>
> cu
>   Reinhard
>
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Re: [Asterisk-Users] Re: newbie ISDN question

[Klaus-Peter Junghanns]

> Is there a list of cards that use this chipset somewhere on the 'net?
> I've googled for it, but most pages only talk about "cards based on the
> HFC-S chipset" without listing brand and model names.
>
Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK DMI-128+
to name a few ;-)

regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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Re: [Asterisk-Users] Multiple phonenumbers on one E1 PRI with Digium TE410P ?

[Klaus-Peter Junghanns]

Hi Jan,

yes you can:

[zap-in]
exten => _49xxx,1,Goto(contextA)
exten => _49xxx,1,Goto(contextB)

regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/

>
> Hi,
>
> one short question: Is it possible for the zaptel driver to deal with
> multiple phone numbers on one single E1 PRI line?
>
> I could make my carrier route +49 xxx a-zzz and +49 xxx b-zzz
> and others down one single PRI trunk to our asterisk box terminating in
> a Digium TE410P.
>
> Does the driver handle this and can I put calls coming in all on the
> same physical interface put into different contexts based on the dialed
> prefix?
>
> Thanks and Regards,
> Jan Baumann
>
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Re: [Asterisk-Users] ISDN p2p AVM Fritz Card

[Klaus-Peter Junghanns]

Hi Stephan,

the passive AVMs do NOT support P2P.

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


Am Mi, 2004-01-21 um 14.39 schrieb Stadlbauer Stephan:
> hello!
> 
> I'd like to use my asterisk-box on a p2p-line from my alcatel pbx. 
> 
> isdn4linux and capi works fine on a s0-line from the pbx, but i need
> more then 8 msns, cause i'd like to implement a voice-mail system and an
> extension to our "old" pbx with voip-phones.
> 
> had anyone success using isdn4linux on a t0 (p2p) line (of course with
> an inexpensive passive card like my avm fritz!, cause i'm still in the
> testing phase, and don't like to buy expensive hardware at this point) ?
> if so - any information would be helpfully.
> 
> by the way i'm from austria - so using euroisdn!
> 
> thanxs in advance, stephan stadlbauer
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Re: [Asterisk-Users] CAPI: Early-B3 working with AVM-B1?

[Klaus-Peter Junghanns]

Hi Karsten,

are you sure your MSN is correct? If not T-Com will replace it
with your "main" MSN and probably will ignore the CLIR setting.

best regards

kapejod
-- 
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


Am Do, 2004-01-22 um 09.00 schrieb Karsten Wemheuer:
> Hi,
> 
> here is an update to my own post to this list.
> 
> Following an information from Philipp, I testet this with an passive AVM
> card, but the same things happen. What am I doing wrong?
> 
> Is there something wrong with my extension.conf?
> 
> without Early B3:
>   exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30)
> with Early B3:
>   exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30)
> 
> 
> Thanks,
> 
> Karsten
> 
> 
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Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?

[Klaus-Peter Junghanns]

hi,

just signed up and it works like a charm. :-)
They even support g711 :) and multiple channels :)

make sure you have in sip.conf:

register => :[EMAIL PROTECTED]/

you will get the too many hops if you try to register
with their proxy (proxy.de.sipgate.net).


best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/



> I set up an account with sipgate yesterday evening and tried to use the
> above mentioned register in sip.conf * to login to sipgate.
> No luck so far.
>
> They use SER and I get "483 too many hops" replies back from them.
>
> Any help is greatly appreciated.
>
> -Walter
>
>
>
> --
>   Walter Doerr   =*=   [EMAIL PROTECTED]   =*=   FAX: +49 2421 962001
>   "The poor folks who only have 100MBytes of RAM five years
> from now may not be able to buffer a 16MB packet, but that's their tough
> luck."  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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Re: [Asterisk-Users] ISDN update

[Klaus-Peter Junghanns]

oh yes...

i added callgroup support for chan_capi. That's why you have to load
res_parking.so before chan_capi.so. So in modules.conf you need.

load => res_parking.so
load => chan_capi.so

[global]
chan_capi.so=yes

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/

>>
>>
> I tried make, make install.
> /usr/bin/asterisk -vvvgc
>
> and what I get is:
> loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so:
> undefined symbol: ast_get_group
> loader.c:358 load_modules: Loading module chan_capi.so failed!
>
> what's wrong?
>
>
>
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[Asterisk-Users] ISDN update

[Klaus-Peter Junghanns]

Hi BRI people,

chan_capi 0.3.1 is now released, including a fix for the pipe leak.

bristuff 0.0.2rc7 is available now too. Including a zaptel driver
for the HFC-S PCI A based ISDN cards (with echo cancelation, TE and
NT mode).
We'll also have a devkit for zaptel BRI soon.

enjoy!

best regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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Re: [Asterisk-Users] Fax

[Klaus-Peter Junghanns]

Hi,

make sure you have echo cancelation disabled on that zaptel
channel.

regards

kapejod
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/

> Hi All,
>
> My asterisk system is running well but I can't send or receive faxes.  I
>  have an analogue fax plugged into a TDM400 connected to my ISDN 2e via
> an Eicon Diva.
>
> I am using G711.U - do I stand a chance of faxing or should I be doing
> it differently?
>
>
> Simon
> --
> Simon Faulkner - Dedicated Programmes
> 01538 303 900 - 07771 845 326
> http://dpnet.co.uk
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Re: [Asterisk-Users] Need to interface to BRIs

[Klaus-Peter Junghanns]

Hi Jim,

we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
You can find more information about it at:
http://www.junghanns.net/asterisk/page17.html

best regards

Klaus
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


Am Mo, 2004-02-16 um 10.10 schrieb Jim Archer:
> Hi All...
> 
> I would like to interface 4 BRI lines to Asterisk.  I looked at Digium's 
> hardware list and, although they have solutions for PRI and T1, I didn't 
> see anything for BRI.  I would like to avoid ISDN4Linux if possible.  Does 
> anyone know of any hardware suppoted by Asterisk I can use for this?
> 
> Thanks
> 
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Re: [Asterisk-Users] Need to interface to BRIs

[Klaus-Peter Junghanns]

Hi Jim,

i forgot to mention that the drivers do not yet support NI-1, but will
support it in the near future. Until then the only solution for you 
will be the Eicon Diva Server 4BRI-8M and chan_capi.

best regards

Klaus
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


Am Mo, 2004-02-16 um 10.53 schrieb Jim Archer:
> I forgot to mention, I am in North America.
> 
> --On Monday, February 16, 2004 4:10 AM -0500 Jim Archer <[EMAIL PROTECTED]> 
> wrote:
> 
> > Hi All...
> >
> > I would like to interface 4 BRI lines to Asterisk.  I looked at Digium's
> > hardware list and, although they have solutions for PRI and T1, I didn't
> > see anything for BRI.  I would like to avoid ISDN4Linux if possible.
> > Does anyone know of any hardware suppoted by Asterisk I can use for this?
> >
> > Thanks
> >
> > ___
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> 
> 
> 
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Re: [Asterisk-Users] Re: Need to interface to BRIs

[Klaus-Peter Junghanns]

Am Mo, 2004-02-16 um 12.49 schrieb Cees de Groot:
> Klaus-Peter Junghanns  <[EMAIL PROTECTED]> said:
> >we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
> >
> One thing I'd like to know about this card: Echo Cancellation? I've
> replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is
> remarkable...
> 
Yes, like any zaptel device it supports echo cancelation (in software).

> (OP: there's also a 4BRI from Eicon, IIRC. It'll work with * through
> CAPI, but I'm quite sure that it's a bit more expensive than KP's card
> ;-))

You can get 2 quadBRI PCI for the price of 1 Eicon 4BRI-8M.


best regards

Klaus
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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Re: [Asterisk-Users] Analogical FXO vs. BRI dialing speed

[Klaus-Peter Junghanns]

Am Mo, 2004-02-16 um 14.39 schrieb Jean-Marc V. Liotier:
> When dialing out, will a call be established significantly faster by an
> ISDN adapter such as an Eicon Diva server compared to an analogical FXO
> such as Digium's X100P ?

Yes, ISDN uses digital signalling so call setup times on the last mile
(from your NT1 to the telco switch) are close to 0. Also the callerID on
incoming calls is available immediately with ISDN (with analog lines you
usually get it after the first ring).
 
best regards

Klaus
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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Re: [Asterisk-Users] Re: Need to interface to BRIs

[Klaus-Peter Junghanns]

The FritzCard has CAPI drivers and does NOT provide zaptel timing.

The quadBRI PCI has zaptel drivers and does provide zaptel timing.


Am Mo, 2004-02-16 um 14.41 schrieb Master Abi:
> Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium 
> Zaptel cards?
> 
> Matteo Brancaleoni wrote:
> > Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto:
> > 
> >>Klaus-Peter Junghanns  <[EMAIL PROTECTED]> said:
> >>
> >>>we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN.
> >>>
> >>
> >>One thing I'd like to know about this card: Echo Cancellation? I've
> >>replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is
> >>remarkable...
> > 
> > 
> > since is zaptel based, it shares same zaptel routines for EC,
> > as far as I know.
> > 
> > Matteo.
> > 
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Re: [Asterisk-Users] chan_capi problem

[Klaus-Peter Junghanns]

Am Di, 2004-02-17 um 09.33 schrieb dfm:
> Hi to all
>  
> I've mada up my mind and i tried to change from i4l to chan_capi,
> following some councelling from the gurus.
>  
> I compiled it up, and when i try to load it in modules.conf, i get
> that wonderful message and Asterisk does not start:
>  
> [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239
> ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined
> symbol: ast_get_group
> Feb 17 09:21:40 WARNING[16384]: loader.c:358 load_modules: Loading
> module chan_capi.so failed!
>  
> Any idea?
>  
> In modules.conf I have:
>  
> noload => chan_modem.so
> load => chan_capi.so
> [global]
> chan_modem.so=no
> chan_capi.so=yes
> 
> But in capi.conf i really don't know what exactly to put, i left it as
> it comes, but i don't know how to set this file up.
>  
> Any one is a chan_capi guru
>  
> Regards
>  
> Diego

here we go again...

put:
load => res_parking.so

before
load => chan_capi.so


best regards

Klaus
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/


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Re: [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode

[Klaus-Peter Junghanns]

Hi Ernst,

use this:
exten => 74341423,1,Dial(Zap/g2/74341423,r)
instead of:
exten => 74341423,1,Dial(Zap/5/74341423,r)

-- 
best regards

Klaus
--
Klaus-Peter Junghanns

CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/



Am Do, 2004-02-19 um 14.24 schrieb Ernst Lehmann:
> Hi, 
> 
> Does anyone operate Asterisk with zaphfc in NT-Mode successfully ??
> 
> I have the problem, that I could not contact my ISDN-Phone on such a
> channel. It rings, but If I pick up the phone, I only get a Hangup in
> the console
> 
> Thanks for any clues on it...
> 
> Here my setup:
> 
> 3 HFC Cards.
> 
> first card is TE, other two are NT mode (loaded zaphfc with modes=6)
> 
> The ISDN-Phone is connected to the second-card. The ISDN-Bus is powerd
> by an old NTBA like described in the pbx4linux project Howto.
> 
> /etc/zaptel.cfg
> 
> 
> span=1,1,3,ccs,ami
> bchan=1-2
> dchan=3
> span=2,0,3,ccs,ami
> bchan=4-5
> dchan=6
> span=3,0,3,ccs,ami
> bchan=7-8
> dchan=9
> ---
> 
> /etc/asterisk/zapata.conf
> 
> -
> switchtype = euroisdn
> signalling = bri_cpe_ptmp
> pridialplan=unknown
> echocancel=yes
> immediate=no
> group = 1
> context=prod
> channel => 1-2
> 
> switchtype = euroisdn
> signalling = bri_net_ptmp
> pridialplan=unknown
> group = 2
> context=prod
> channel => 4-5
> -
> 
> valid passage from extension.conf
> 
> --
> [prod]
> 
> 
> exten => 74341423,1,Dial(Zap/5/74341423,r)
> 
> .
> 
> ---
> 
> The ISDN-Phone is configured to listen on the MSN 74341423
> 
> Here the log from the console:
> 
> ---
> 
> -- Executing Dial("Zap/2-1", "Zap/5/74341423|r") in new stack
> -- Called 5/74341423
> -- Hungup 'Zap/5-1'
> -- Accepting call from '8974341421' to '74341423' on channel 2, span
> 1
> MFE for TEI = 64
> -- Timeout on Zap/2-1
>   == CDR updated on Zap/2-1
> -- Executing Goto("Zap/2-1", "#|1") in new stack
> -- Goto (prod,#,1)
> -- Sent into invalid extension '#' in context 'prod' on Zap/2-1
> -- Executing SetVar("Zap/2-1", "starttime=1077196512") in new stack
> -- Executing Playtones("Zap/2-1", "info") in new stack
> -- Executing Wait("Zap/2-1", "1") in new stack
> -- Executing Playback("Zap/2-1", "invalid") in new stack
> -- Playing 'invalid' (language 'en')
> -- Executing Wait("Zap/2-1", "1") in new stack
> -- Executing GotoIf("Zap/2-1", "1?2:7") in new stack
> -- Goto (prod,i,2)
> -- Executing Playtones("Zap/2-1", "info") in new stack
> -- Executing Wait("Zap/2-1", "1") in new stack
> -- Executing Playback("Zap/2-1", "invalid") in new stack
> -- Playing 'invalid' (language 'en')
> -- Channel 2, span 1 got hangup
>   == Spawn extension (prod, i, 4) exited non-zero on 'Zap/2-1'
> -- Hungup 'Zap/2-1'
> 
> -
> 
> perhaps helpfull... some information from dmesg on loading of modules:
> 
> -
> Zapata Telephony Interface Registered on major 196
> zaphfc: start
> PCI: Found IRQ 5 for device 00:09.0
> PCI: Sharing IRQ 5 with 00:04.3
> zaphfc: card configured at mem 0xc8875000 fifo 0xc70e8000(0x70e8000) IRQ
> 5 HZ 10
> 0
> zaphfc: ZTHFC1/0/1
> zaphfc: ZTHFC1/0/2
> zaphfc: ZTHFC1/0/3
> zaphfc: registered zaptel device!
> zaphfc: Card 0 configured for TE mode
> zaphfc: resetting card.
> zaphfc: layer 1 state = F4
> PCI: Found IRQ 12 for device 00:0a.0
> zaphfc: card configured at mem 0xc8877000 fifo 0xc7158000(0x7158000) IRQ
> 12 HZ 1
> 00
> zaphfc: ZTHFC2/0/1
> zaphfc: ZTHFC2/0/2
> zaphfc: ZTHFC2/0/3
> zaphfc: registered zaptel device!
> zaphfc: Card 1 configured for NT mode
> zaphfc: resetting card.
> zaphfc: layer 1 state = F5
> zaphfc: layer 1 state = F6
> zaphfc: layer 1 state = G2
> PCI: Found IRQ 10 for device 00:0b.0
> PCI: Sharing IRQ 10 with 00:11.0
> zaphfc: card configured at mem 0xc8879000 fifo 0xc7148000(0x7148000) IRQ
> 10 HZ 1
> 00
> zaphfc: ZTHFC3/0/1
> zaphfc: ZTHFC3/0/2
> zaphfc: ZTHFC3/0/3
> zaphfc: registered zaptel device!
> zaphfc: Card 2 configured for NT mode
> zaphfc: resetting card.
> zaphfc: bchan rx fifo not enough bytes to receive! (z1=518, z2=8191)
> zaphfc: bchan rx fifo not enough bytes to receive! (z1=518, z2=8191)
> zaphfc: 3 card(s) in this box.
> Registered tone zone 3 (Netherlands)
> zaphfc: layer 1 state = G3
> 
> ---


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