points didn't respond with something Asterisk
could use for some reason.
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Tais M. Hansen
ComX Networks A/S
Tel: +45-70257474
Fax: +45-70257374
pgp9dl9fKymkh.pgp
Description: PGP signature
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ople that the Digium card
would perform better in a 133MHz slot seems quite odd to me.
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ComX Networks A/S
Tel: +45-70257474
Fax: +45-70257374
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Description: PGP signature
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Ast
can't really see how a 133MHz
slot would make any difference?
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Tel: +45-70257474
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e = myrealm
context = incoming
realm = myrealm
dtmfmode= rfc2833
language= da
nat = yes
callgroup = 20
pickupgroup = 20
callerid= "SNOM" <202>
qualify = 3000
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Tais M. Hansen
ComX Networks A/S
Tel: +45-70257474
Fax: +45-70257374
pgp6
0P is just a flakey card.
> Anyone got a DL360 G3 going with a TE410P and FC3?
I did manage to get a TE110P running on the DL380 G4. Still can't get the
TE410P working in the G4 though. Supports your theory.
Sadly we're now being forced to look elsewhere for PRI cards.
--
Reg
kind of status
or something from the card. Any indicator of whats going on?
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ComX Networks A/S
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ork
on the G4.
Our G4 config is dual xeon 3.6ghz, 2gb ram, kernel 2.6.10 and 2.4.28.
zaptel and wct4xxp modules loads fine. At this point the flashing red lights
on the wct4xxp are turned off. zttool shows all spans are OK, no matter if
there are anything plugged in.
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Tais M.
efore the function in which the inline
functions are called (line 810-814).
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Tais M. Hansen
ComX Networks A/S
Tel: +45-70257474
Fax: +45-70257374
pgpnjfUbNjm2x.pgp
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[EMAIL PROTECT
source call# 16385, timestamp
15ms AUTHREQ
11:54:00.955938 a.a.a.a -> b.b.b.b IAX2 IAX, source call# 12, timestamp 0ms
INVAL
11:54:10.956181 a.a.a.a -> b.b.b.b IAX2 IAX, source call# 12, timestamp
10009ms LAGRQ
11:54:10.956326 b.b.b.b -> a.a.a.a IAX2 IAX, source call# 1, timestamp 0ms
I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Any reason why pickupgroup has been limited to 31? 31 groups are quickly used
up when you have multiple companies on the same server.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE
fixes the problem.
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Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFA1tdg32si/nlrQ5gRAgtZAJ44hTH7MU5d5nm3WxLf3oU5eIBlMgCeObbB
tuMz2l1pnlqtsAwgL9xER04=
=1pZQ
-END PGP SIGNATURE-
Index: apps
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 15 June 2004 14:59, Luckcuck Nick-LCKN001 wrote:
> CFLAGS+=-fPIC
> CFLAGS+=-I../asterisk
> CFLAGS+=-I/usr/include
Don't know if it helps, but try adding:
CFLAGS+=-I../asterisk/include
- --
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Tais M. Hansen
ComX Net
ime=895183800
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFArcap2TEAILET3McRAiO5AJ4uVuolDW6/V6O6zMDOIok+kirI1wCfXoSq
yezJIP1aSCIEUBKmQtTTW4s=
ing IAX2 out of play, the timestamps seems to be just fine.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFAq0dY2TEAILET3McRAvt2AKCh1AHED6sSmsfEhsRTQ8DKJMRE+gCfXsyV
wRw5X3aHtWzbleycJGx5yRU=
=m3
ia IAX2 and got seriously distorted sound. One server had
speex-1.1.4, the other speex-1.1.5.
Also, I've been unable to get below 20 for speex on a Xeon 3.02ghz.
- --
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v
seems to be disabling EC on the same channel twice?
(CVS-04/09/04-23:00:24-CEST)
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFAh8CM2TEAILET3McRAuRyAJ9cNxBXuBVKVbWVYJHBRSf/qVPzjQCcDjDG
e as well today. So I tried the latest zaptel version
which is currently locking up my system. I'm currently trying to get some
debug output from zaptel/wct4xxp for one of the Asterisk devs.
- --
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
using these problems.
I used the following in the zaptel cvs directory to roll back the zaptel
sources:
cvs up -D "2004-03-05 10:28"
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8D
ar 30 12:01:41 NOTICE[98311]: chan_zap.c:4933 handle_init_event: Alarm
cleared on channel 61
Mar 30 12:01:41 NOTICE[98311]: chan_zap.c:4933 handle_init_event: Alarm
cleared on channel 62
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGN
ort for
> same codecs in the same order. Ie.
I have:
disallow=all
allow=alaw
> If you are using one of the latest versions of * (not sure exactly which
> one), IAX and IAX2 have different configuration files.
I have only used IAX2... I was thinking that maybe Transfer() needs IAX-IA
quot; in your Dial statement? i.e.
> Exten => someextension,1,Dial(IAX2/SOMETHING,20,t)
Yes, I've tried with both 't' and 'T'.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU
ll to a different
extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing
happens and after a few seconds, the line is hung up.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/
nd useful to some people.
I didn't know SIP or IAX actually defined causes for hangups. I any case one
of us should make this a feature request on bugs.digium.com.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.
ould use my approach and combine it with the CDR userfield. Personally I
would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFAGn8E2TEA
5,Hangup()
exten => c4,1,Congestion()
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFAGlKy2TEAILET3McRAv7gAKCREpAN3kVvbEuTDAQkU9kb6IrZiQCdEXlR
3FroTgPgWQmBrq
tly never working ${HANGUPCAUSE}
> variable)?
HANGUPCAUSE is working fine here (cvs).
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQFAGkv82TEAILET3McRAjGcAJ9FzGmcXX8jJwjs30hVjhAO3pcO5ACfZ
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 23 January 2004 15:08, Tais M. Hansen wrote:
> How can I configure the TE410P card to act as master instead of slave?
Ah, the wiring was wrong. Straight when connected to the Telco, crossover when
connected to a local PBX.
- --
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
How can I configure the TE410P card to act as master instead of slave?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux
ut when i try to load the module it gives
> the following errors:
> #modprobe zaptelO.k
> # modprobe wct1xxp gives
You need to use wct4xxp for the TE410.
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ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE--
t then those get wiped out
> upon bridging. Adding the "/n" option prevents this.
This certainly solved one half of my problem! Thanks!
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
ck
A matching segment of the dialplan:
exten => s, 2,NoOp(id = ${id})
exten => s, 3,Answer()
exten => s, 4,NoOp(id = ${id})
exten => s, 5,Wait(1)
exten => s, 6,NoOp(id = ${id})
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-
3" in thesecond expression change things? Or is that
> outside the scope of what you're asking?
It says, if true goto priority 3. Outside the scope. :)
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Ver
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 December 2003 13:08, Tais M. Hansen wrote:
> I'm having a problem with the following expression examples.
> exten => s,1,NoOp($[$[${value} >= 10] & $[${value} < 18]])
> exten => s,1,GotoIf($[$[${value
examples above. First extension evaluates to 1 while
second evaluates to 0 even though it's the same expression used in both
examples. Am I doing something wrong?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Versio
#x27; in the url above.
http://almaw.com/ethereal-iax2/
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQE/zI882TEAILET3McRAvSRAJ94JwTJlRv6Sjj/3aq5YQE9sIwJqwCfRuUO
EG1KvYQsjmuT+2zyKYLFrjg=
=fM0n
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 02 December 2003 14:03, Alastair Maw wrote:
> > Could you please create a URL that is a bit more non-version-specific?
> http://almaw.com/etheral-iax2/
> It now, inevitably, has a web site. :)
Giving a 404. :)
- --
Reg
fresh install.
I'll back that up. It's all Slack here :)
The only problem we've experienced is with the g729 codec which is currently
still unusable (makes Asterisk segfault)
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE---
the inband
tone and the RFC2833 tone, which will result in two tones.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQE/vNa12TEAILET3McRAv/nAKCFrJcOpp8TsPpHUZrcOxL8LILoRACgj0tz
fOJ6graDy
In the setup below we do not experience double DTMF from SIP to PSTN but DTMF
is still squelched from PSTN to SIP. SIP is configured for rfc2833 on both
the endpoint and on Asterisk in this example.
SIP > Asterisk1 > ZAP > PSTN
It sounds to me like there's a bug here somewhere?
es or tests yet.
Before I start to analyse this periodic problem, I thought I'd just check with
the list and see if anyone has had a similar experience?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GN
ur serverpark is based on the same
hardware all the way.
But this is not the issue here. I've already contacted Digium about the irq
issue and the E400P issue.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG
ad
Nov 10 06:00:48 gw-4 kernel: TE410P: Double/missed interrupt detected
Although /proc/interrupts doesn't show any errors or misses.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQE/r51T2TE
cards (because it's a
sip/iax2 gateway) then you're screwed?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQE/r5Ah2TEAILET3McRAoGnAKCiY1d63hTFhfyUBgWCy2NmyFQ0mQCeKCWo
aoUmaCA0YeHvgcL
gards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQE/lURd2TEAILET3McRAkBmAJ4vva/kqMFLFiRHRUHyIegh4Ml7AACfVcK9
zbrwPYjgTK0EAoJ/iiT1cIs=
=+Lz5
-END PGP SIGN
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
How do I prevent call forward looping over multiple Asterisk systems?
I.e.
SIP1---Ast1===Ast2---SIP2
SIP1 forwards his calls to SIP2 and SIP2 forwards to SIP1.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
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he phone
receiving the call wouldn't see the actual caller.
Any way to work around this?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
iD8DBQE/iWrb2TEAILET3McRAqSQAJ4rhu1xwd0InXtHE8RpHwl6hKv
times that of the codec bitrate (gsm) just seemed wrong to me.
I haven't yet had the chance to test multiple calls through IAX, so I hope
you're right. :)
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
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-BEGIN PGP SIGNATURE-
Version: GnuPG
-e1pri/telco
The measured values came from nic2 of Asterisk1 by having iptraf measuring the
iax port 4569 using "Statistical Breakdown by TCP/UDP port" after having
added the port to the portrange in iptraf config. Promiscuous mode off.
I'm quite sure nothing but the iax udp tra
ding codec definition)
gsm: 52 kbps (13 kpbs)
alaw: 154 kbps (?)
speex: 57 kpbs (24 kpbs)
Seems a little high to me?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux
UST be released under GPL,
> unless you obtain a license to release them outside of GPL from Digium.
Just to be on the safe side: What if the module is not released publically but
is used in production systems internally?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-7
; http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+channels
> Do you think we've missed your point there? If so, help us correct this
> problem. All pages in the wiki are open for editing.
Nice page.
I've added a comment. Don't know if my explanation is any good though.
"[xxx]" entries.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)
iD8DBQE/caz82TEAILET3McRAvD6AJ98K0jGtZLWTFh/OdjpGHxyAgz71gCfZIXI
LbIVeURHr1OFLqQ/GEEZESU=
=/k8y
-E
talk to Grandstream 100s every day, and also to the FWD
> network without issue. Are you trying to access SIP across NAT or other
> restrictive firewall or something?
No firewall, no nat. Only lack of documentation for sip.conf... I got it to
work now.
- --
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Tais M. Hansen
ComX Netw
archives as there are sample configs
> in there somewhere.
I did, but none of the examples nor where the default config clear on the fact
that the "[xxx]" HAD to match "sip:[EMAIL PROTECTED]". Once that was cleared up, it
started working for me too.
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Ta
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Now that I've been unable to register 2 hardware SIP phones and one software
(Kphone), I'm beginning to doubt that chan_sip works at all.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP
ommand (fd=153, s=0x0) at cli.c:1006
#12 0x08085770 in netconsole (vconsole=0x80bcc28) at asterisk.c:193
#13 0x40023463 in pthread_detach () from /lib/libpthread.so.0
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2
ten => 1234,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]|30|r)
calls for 30 seconds, but
exten => 1234,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED],30,r)
only calls for 3-5 seconds and then hangs up as if nobody answered.
- --
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Tais M. Hansen
ComX Networks
Tel: +45-702
dication on my
h323 endpoint that it's actually ringing.
Using chan_oh323 instead of the native h323, the problem magically disappears.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)
iD8DBQE/brqt2
0|r) ... Works fine with OH323 though.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)
iD8DBQE/bq4/2TEAILET3McRAmUkAJ97wvMuj+O5D1E3Uu9PBZ5G4bIt+QCgiFAA
Ufbc9xNFKXOExxoXia0Qits=
=iRbu
-END
exceeded on call [hex]@
as.as.as.as for seqno 102 (Request)
Sep 19 11:09:04 WARNING[147466]: File chan_sip.c, Line 443 (retrans_pkt):
Maximum retries exceeded on call [hex]@
as.as.as.as for seqno 102 (Request)
What on earth is this? Codec?
- --
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Tais M. Hansen
ComX Networks
Tel: +45-702574
is all about ?
Pretty straight forward. A SIP phone at '192.168.1.70' failed registration at
your Asterisk box at '192.168.1.1'.
Try "sip debug" at your CLI, and you'll see similar messages as the ones I
described in my "SIP registration" thread.
eral]
port=5060
bindaddr=s.s.s.s
context=cxnet-in
tos=lowdelay
[siptestphone]
type=friend
user=atrg613test
host=dynamic
defaultip=c.c.c.c
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 15 September 2003 19:49, Tais M. Hansen wrote:
> Before I hack out the ',' -> '|' tr in extension.conf parser, any way to
> escape ',' that I missed?
Apparently not.
Hacked. Patch will be posted
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Before I hack out the ',' -> '|' tr in extension.conf parser, any way to
escape ',' that I missed?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
learly defined
eventually when the government figures out their version of the
anti-terror-package.
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)
iD8DBQE/YYZK2TEAILET3McRAvaDAJ9a8avQeKRqj/5TUzXiE787
on enables restrictcid which seems to hide the callerid on
outgoing zap calls. I haven't had the opportunity yet to test what happens on
other channel types.
*patch disclaimed*
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 09 September 2003 17:54, Tais M. Hansen wrote:
> How do I optionally hide the caller id on outgoing calls on chan_zap? Ie.
> calling h323 -> asterisk -> chan_zap -> isdn provider.
Problem solved. I made app_dial.c take an o
ne has to send something along with the callerid
(set a flag in the call setup or similar) to indicate that the callerid is
private.
Any pointers?
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/L
s limited in the source code (pbx.c).
Line 67 in include/asterisk/cdr.h
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Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)
iD8DBQE/XF6N2TEAILET3McRAhztAJwPip5Sb0Xv9QzZe9uzk9XoEBoXGgCcCRjK
uHBForE5bJUUVg
() from /lib/libc.so.6
Does it require anything in the kernel that I might accidently have switched
off?
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.2 (GNU/Linux)
iD8DBQE/VaYE2TEAILET3McRAh8yAJ4h0d8ywMUE2LMT8AB5Ber/2UBU7QCeOA36
rMLOqtIQHWg374TPu10TDX0=
=Ri9s
-EN
6
User input mode: 2
Max number of inbound H.323 calls: 1000
Max number of outbound H.323 calls: 1000
Box: Intel Xeon 2.4GHz, 533 FSB, 1GB RAM.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.2 (GNU/Linux)
i
t; suddenly not generate CDRs?
> The CDRs are generated by Asterisk. It is not something that
> is controlled by the channel driver.
Well, whatever was causing this seems to be gone with latest asterisk cvs and
oh323 0.5.5.
Thanks all.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +4
your system setup?
I'm using the G711ulaw codec. I experienced pops making oh323 -> zap calls and
oh323 -> oh323. Both with and without a gatekeeper. If there's anything
specific you need, please let me know.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 21 August 2003 07:02, Manoj K Gupta wrote:
> you can also try setting AMAflag=billing in oh323.conf if it helps.
Already tried that. Didn't make a difference. I'll test again with the latest
oh323 (0.5.5).
- --
Regards,
T
call created a CDR record. And a zap -> oh323 call also
inserts a CDR. Strange. What could cause oh323 initiated calls suddenly not
generate CDRs?
I'm currently not using cdr_csv's.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGI
ved pri dchannel debug and hid phone and ipnumbers.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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iD8DBQE/QQ3e2TEAILET3McRAgpKAJ90VNuxjtCZOVugUXqjG9PCOXaTygCfTWw3
tpp9PQJdJDRk6Vm9ZdagzAw=
=obWn
-EN
I suppose. I'm getting a "Successfully connected to MySQL
database." But it still doesn't insert CDRs.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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iD8DBQE/QQu/2TEAILE
...
That's not an option. CDR's needs to be 101% reliable in our environment. :(
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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A
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 18 August 2003 18:19, Tilghman Lesher wrote:
> On Monday 18 August 2003 11:08 am, Tais M. Hansen wrote:
> > Is cdr_mysql broken in latest CVS? It builds and loads fine but it
> > doesn't insert cdrs in the database
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Hi,
Is cdr_mysql broken in latest CVS? It builds and loads fine but it doesn't
insert cdrs in the database and there's no debug output at all.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP
from h323 to oh323 recently partyl because things like
ringing-while-calling didn't work with h323. Also, h323 didn't support "*" as
a gateway prefix whereas oh323 does, which I needed in one test setup.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi.
Using inAccess Networks chan_oh323, I'm experiencing some clicks or pops, how
can I fix that?
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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On Thursday 31 July 2003 16:52, Tais M. Hansen wrote:
> I place a call via one ISDN carrier (or from my cellphone) through another
> ISDN carrier via chan_zap into Asterisk, which then tries to dial the H323
> phone. But the calling ph
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On Tuesday 29 July 2003 13:47, Tais M. Hansen wrote:
> I havn't used the h323 channel of Asterisk for a while, but today I needed
> to test a few things only I found out that Asterisk/H323 crashes my Siemens
> optipoint 400 phone. It s
one. But the calling phone keeps ringing after congestion has been called.
Calling Busy() instead of Congestion() displays the same behaviour.
It seems to work though when I do something like the following:
exten => s,1,Dial(H323/ip$${ARG1}|15)
exten => s,2,Answer()
exten => s,3,Congesti
ot;) in new stack
... But the calling phone just keeps ringing.
- --
Regards,
Tais M. Hansen
ComX
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/1hoH7dj5sZ5b0f2Rmb3cA=
ng. Anyway to get around it?
- --
Regards,
Tais M. Hansen
ComX
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iD8DBQE/J5nT2TEAILET3McRAnprAJ42dxXWweL9RWGBKd0DJDwuV6RBHgCgiO0A
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=dYqc
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___
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Hi,
Why do the string specified with SetCIDName() get overwritten with the name of
the user running Asterisk, when placing calls through H323?
- --
Regards,
Tais M. Hansen
ComX
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ing broken in chan_h323?
- --
Regards,
Tais M. Hansen
ComX
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iD8DBQE/Jl7g2TEAILET3McRAmTzAJwNCGxGKJI5ZK92Vnri0CUJM+hFpQCfZo5o
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=8xpI
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__
d give me was two ETSI
document references, which I didn't find very usefull. And to top it off, I
asked them to confirm that they had actually connected the ISDN30 in their
end, which they suddenly realised a few minutes ago, that they hadn't. :(
- --
Regards,
Tais M. Hansen
ComX
--
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Does anyone know how to configure the spans on the E400P for use with "ISDN-E
DSS1" (Ericsson FS standard ISDN30)?
- --
Regards,
Tais M. Hansen
ComX
-BEGIN PGP SIGNATURE-
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me a PDF of it ??
Is this what you're looking for? :)
http://www.digium.com/downloads/quick_install_zaptel_asterisk.pdf
- --
Regards,
Tais M. Hansen
ComX
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iD8DBQE/AC0z2TEAILET3McRAij4AJ4z9a09G8eBIwjD76mHQhtKnH/aNQCdGvNe
LTc4x6WPjv
I've seen this one too. Usually right after an alarm has been raised or
cleared.
- --
Regards,
Tais M. Hansen
ComX
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iD8DBQE+6bU82TEAILET3McRAvtsAJwPRCXgvzzP1J2D081uZlalx/TgJgCfUXOO
g1JSr5lShkkNuLr0LnPeOMI=
=1j
isk
to an h323 phone disappears in thin air. The easy (but ugly) way to solve
this is to add a 3 second wait after answer.
Is there any other fix for this? Something in the works?
- --
Regards,
Tais M. Hansen
ComX
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Q.SIG, and everyone will benefit. :-)
I'd love to try, if I had any clue on how. I took a quick peek at the libpri
sources and I doubt I'll figure it out anytime soon. :(
- --
Regards,
Tais M. Hansen
ComX
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iD8DB
-BEGIN PGP SIGNED MESSAGE-
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On Tuesday 03 June 2003 12:52, Tais M. Hansen wrote:
> The Siemens side is configured to "ISDN30: ECMA QSIG". The Siemens manual
> states the card provides 30 ISDN B-channels which can be used for trunking
> or networking.
Okay,
ove info? Can it be
done? I'm pretty lost.
- --
Regards,
Tais M. Hansen
ComX
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=GfQa
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__
behaviour.
- --
Regards,
Tais M. Hansen
ComX
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=wWFc
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Index: ztcfg.c
==
ata, ppp_generic and slhc and modprobing tor2 again. Same result.
Build-out does change as expected when changing the x in span=1,0,x,... It's
only the d4/ami that puzzles me.
- --
Regards,
Tais M. Hansen
ComX
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iD8DBQE+22NV2TEAIL
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