[Asterisk-Users] Ubuntu Migration

[Matthew Walster]

I've just migrated Asterisk from my old Gentoo system to an Ubuntu system, 
copied across all the /etc/asterisk files and now it fails to work. After 
brief looks, I find that it can't access:

/var/log/asterisk/messages
/var/run/asterisk.ctl
/var/run/asterisk.pid

So I touched these files, and chown/chgrp'd then to user/group 
asterisk/asterisk. Now, when I run asterisk through /etc/init.d/asterisk 
(Ubuntu is a Debianalike) it says Unable to set high priority.

Any ideas?

Cheers,

Matthew Walster


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Re: [Asterisk-Users] Asterisk - Spandsp: fax header

[Peter Svensson]

On Wed, 18 May 2005, Steve Underwood wrote:

 The header is always in the received image. The TIFF file contains 
 exactly the same image that a receiving FAX machine would print out.

I think he is refering to the remote fax id to be presented, not the 
header. I.e. the 20 digit user selectable number on the remote fax. The 
one often seen on the lcd of the receiving fax and so on.

Peter


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Re: [Asterisk-Users] Ubuntu Migration

[snacktime]

On 5/17/05, Matthew Walster [EMAIL PROTECTED] wrote:
 I've just migrated Asterisk from my old Gentoo system to an Ubuntu system,
 copied across all the /etc/asterisk files and now it fails to work. After
 brief looks, I find that it can't access:
 
 /var/log/asterisk/messages
 /var/run/asterisk.ctl
 /var/run/asterisk.pid
 
 So I touched these files, and chown/chgrp'd then to user/group
 asterisk/asterisk. Now, when I run asterisk through /etc/init.d/asterisk
 (Ubuntu is a Debianalike) it says Unable to set high priority.

Debian has it's own way of installing asterisk.  You should probably
install asterisk again, then copy over only the files you need from
your gentoo box instead of copying the whole directory over.

Or you can install from source which is the best way IMO.

Chris
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Re: [Asterisk-Users] VoiPSupply Dot Com

[gsmith]

All:

First let me thank everyone for the good words. It is much appreciated by
all of us at VoIPSupply.com. All of our numbers are up and working. There
are instances from time to time, when T's or PRI go down and we are
without phones services for a few minutes, but this is always kept to a
minimum.

If there was a problem it certainly was not on our side. There is now way
we would go an entire day without our toll free numbers working. This is
one of our life lines.

To those who are wary of purchasing from VoIPSupply.com, we offer the
following information to you. We have been in business since 2002. We are
North America largest VoIP Equipment VAR. We are currently generating over
1 million dollars per month in revenue. We are an industry leader. We are
here for the long haul.

We do not ship COD, as it is does not exist for international shipments. I
for one have done a large amount of business with the Atserisk community.
I am constantly offering specials, and discounts to the community, above
and beyond our low web prices.

It is sad to hear that you will not be purchasing from us. I do not
understand though, why we owe you an explanation for our toll free number
being down.


Lastly, we do charge for technical support. We are hear to help, but the
low margins on ATA's etc certainly does not leave us room to give away
free support. All of you that are ITSP's know exactly what I am talking
about.


If you order something, and you can't get it to work, you can pay for us
to make it work for you. If you order the wrong product, then that is your
mistake not ours.

There is an open invite to all to call or email me at any time to discuss
or business. Constructive criticism is always welcomed.

Thank you all for business and we look for more in the future!

Garrett Smith
VoIPSupply.com
[EMAIL PROTECTED]
716-250-3408 Direct


 mr. barker wrote:

 I tried calling their toll free number and toll number last week in
 the morning and afternoon and was handed a recording saying this
 number is no longer in service. The web site was up but there was no
 message on the site as to why the phone numbers were not working.

 I just called the number now and it is working.

 Being around the internet for a quite a long time this gives me an
 uneasy feeling. I have seen company’s start to go under and pull the
 plug when they get into financial trouble(not being able to pay the
 bills) and run with the customers money. I have had this happen to me
 on 2 occasions. Just the woes of doing business on the net.

 Being in Canada it makes it very difficult to find companies that will
 ship COD from the US. If I was to order I would only order COD from
 now on from VoipSupply.

 I have ordered product from VoipSupply and received the product. I
 will not be ordering more product do to this outage of the phones with
 no explanation.

 Just my 2cents.

 Maybe the tollfree provider was responsible for the outage and maybe it
 only affected service from Canada.

 They accept credit cards and paypal. I believe you would have some
 recourse if they ran with your money.

 I quit shipping anything COD to anywhere a few years ago. If the
 customer refuses delivery the vendor loses money. When UPS instituted a
 policy of not handling cash payment for COD, I quit for good.

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Re: [Asterisk-Users] VoipSupply.com

[gsmith]

JD:

Your are correct. B2 Technologies is our parent company.

Thanks,

Garrett

 I've ordered several things from them; all arrived as expected.
 Last time I ordered from voipsupply but the order was fulfilled by B2
 TECHNOLOGIES LLC (same company I think).

 JD

 Manjit Riat wrote:

 I am going to buy some IP phones from them but I sent them an email
 couple of weeks ago and got no reply. Has anyone ordered anything from
 them? Any other places that I can buy from? Sorry if it's a wrong post.



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 --
 JD Austin
 Twin Geckos Technology Services LLC
 email: [EMAIL PROTECTED]
 http://www.twingeckos.com
 phone/fax: 480.422.1250

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Re: [Asterisk-Users] Ubuntu Migration

[Matthew Walster]

On Wednesday 18 May 2005 07:15, snacktime wrote:
 Debian has it's own way of installing asterisk.  You should probably
 install asterisk again, then copy over only the files you need from
 your gentoo box instead of copying the whole directory over.

The only files I've changed are extensions.conf and sip.conf - the rest are 
left as they are.

Just a quick listing:

[ 20050518T075145BST | [EMAIL PROTECTED] ] ( /etc/asterisk )
548 $ ls
adsi.conf   enum.confmeetme.conf  res_config_odbc.conf
adtranvofr.conf extconfig.conf   mgcp.conf res_odbc.conf
agents.conf extensions.conf  modem.confrpt.conf
alarmreceiver.conf  extensions.conf.old  modules.conf  rtp.conf
alsa.conf   features.confmusiconhold.conf  sip.conf
asterisk.adsi   festival.confosp.conf  sip.conf.old
asterisk.conf   iax.conf oss.conf  skinny.conf
cdr_manager.confiaxprov.conf parking.conf  telcordia-1.adsi
cdr_odbc.conf   indications.conf phone.confvoicemail.conf
cdr_pgsql.conf  logger.conf  privacy.conf  vpb.conf
cdr_tds.confmanager.conf queues.conf   zapata.conf

So, if I just copy extensions.conf and sip.conf, the Debian one will work?


 Or you can install from source which is the best way IMO.

I'm sorely tempted... If you want to do my Java coursework (due in a few 
days!) then sure! Otherwise, I'll just get it up and running first then do it 
the proper way =)

Cheers,

Matthew Walster
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RE: [Asterisk-Users] Background() problem (with queue(), etc.)

[Adam Goryachev]

On Tue, 2005-05-17 at 17:04 +0100, Seb Auriol wrote:
 In fact, this is what I'm doing at the moment on the production system, but
 we've had a complaint because it doesn't start at the beginning for each
 caller. This is pretty important because the file starts with something like
 Thank you for calling X. We appreciate your patience during this brief
 period...
 
 Thanks for the info on Background though. I think the wiki could do with
 some clarification on this.
 
 Kind regards,
 

Search the wiki for alternate MoH options, I think there is at least one
which will support playing the music from the beginning for each caller.

Also check bugs.digium.com I suppose...

Regards,
Adam


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Re: [Asterisk-Users] Ubuntu Migration

[Matthew Walster]

On Wednesday 18 May 2005 07:15, snacktime wrote:
 Debian has it's own way of installing asterisk.  You should probably
 install asterisk again, then copy over only the files you need from
 your gentoo box instead of copying the whole directory over.

Oh. Dear God. I just did apt-get remove asterisk. Little did I realise, my 
subconscious had other plans and put --purge in there.

My asterisk config folder just disappeared.

How p*ssed am I?

Matthew Walster


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[Asterisk-Users] DEBUG output on sip extensions

[Marty Mastera]

Can anyone help me 
to understand what the significance of this output is?

May 17 10:50:23 
DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4May 17 10:50:23 
DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and 
SIP/outbound-7dc3

I searchedfor 
these phrasesbut am coming up short on what they really mean. I'm 
trying to investigate problems we are having with two separate asterisk 
installations both using Polycom IP-500 phones. These type of messages 
appear in the logs of both servers. It almost appears as though these 
messages are normal following completion of a call (a hangup), but we are 
troubleshooting bad audio in both locations and the wording of these messages 
doesn't appear benign.

thanks

Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX: 
206.666.1786

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Re: [Asterisk-Users] multiple sip accounts from same sip registrar

[Peter Bowyer]

On 17/05/05, Matt Scott [EMAIL PROTECTED] wrote:
 Dear all,
  
 I have an asterisk sip issue which I don't believe is unique.
 I use a registrar (sipgate.co.uk) where I have 3 different accounts.
 These accounts provide me with three seperate local phone numbers which
 allow me to allocate them to seperate users.
 By using just one of these accounts I can set asterisk up to send and
 receive calls no problem.
 However, when I start to introduce an additional account I start to run into
 problems.
  
 if I do a 'sip show peers' with a good config I think it may outline the
 problem
  
 sip show peers
 Name/username  HostDyn Nat ACL Mask Port
 Status
 1005/1005  (Unspecified)D  255.255.255.255  0   
 Unmonitored
 1004/1004  (Unspecified)D  255.255.255.255  0   
 Unmonitored
 1003/1003  (Unspecified)D  255.255.255.255  0   
 Unmonitored
 1002/1002  10.0.0.52D  255.255.255.255  5060
 Unmonitored
 1001/1001  10.0.0.51D  255.255.255.255  5060
 Unmonitored
 sipgate1/321   217.10.79.219N  255.255.255.255  5060
 OK (52 ms)

I'm not sure what you think the problem is, you haven't told us... but
anyway, I haven't succeeded in sending sipgate inbound calls through
separate contexts, but I deal with them all in a single context - the
calls will arrive at an extension matching the individual sipgate
username in the register command.

Works for me and several others

Peter


-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] VoipSupply.com

[kyle Hagan]

I can get you New 7960's for $299.99 each + Shipping or Refurb for 
$259.99 each plus shipping.
Can get better prices for qty discount. Which Polycoms are you looking for?

Email me off list
Kyle
[EMAIL PROTECTED]

Manjit Riat wrote:
Looking for 7960s and a few Polycom IP300s and IP600s
Have heard great things about IP600. I hope IP300 is also as great as IP600.
Thanks for your replies.
-Original Message-
From: Kyle Hagan [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, May 17, 2005 5:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoipSupply.com

Manjit Riat wrote:
 

I am going to buy some IP phones from them but I sent them an email 
couple of weeks ago and got no reply. Has anyone ordered anything from 
them? Any other places that I can buy from? Sorry if it's a wrong post.

   

I have used them many times and had no problems.
What are you looking for?
Kyle

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Re: [Asterisk-Users] 2 minutes pause before ring on H323 channel

[John Daragon]

Peter Valkov wrote:
John Daragon wrote:
   

Peter Valkov wrote:
 

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support
as described in README file.
Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2
kernel-2.6.11
I tested it with following phones: -- XLite (SIP softphone)
-- QMix SIP IP phone (PA168F)
-- SJPhone (H323 softphone)
-- QMix H323 IP phone (PA168F)
-- FireFly (IAX2 softphone)
Everything works fine except a problem with h323 extension dialing.
Behavior is the same for both
SJPhone (soft phone) and QMix (PA168F). When I dial such extension I
have to wait 2 minutes
exactly (120 seconds) before extension rings. After long way of trial
and errors with .conf files
I managed to minimize this time to 1 minute exactly (60 seconds)
exten = 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause
before ring exten = 21,1,Dial(H323/h323phone at 192.168.0.101) ; this
leads to 60 seconds pause before ring
   

Peter, hi;
I haven't looked at the openh323 code, and I might not get time to...
but in my limited experience, 60 second delays are almost always DNS
timeouts.
 

Yep - down in openh323/src/transports.cxx there's a method
H323TransportAddress::GetIpAndPorts() which is called (eventually) by
MakeCallLocked().  This in turn calls GetPortByService() and
GetHostByAddress().
My guess is that the 60 second wait is caused by a request to a DNS
server that is never honoured.
Of course, I've been wrong before...
   

It is definitely DNS problem. The strange thing is that from command line
everything works just fine. I can perform DNS and reverse DNS lookup without
problem.
Here follows my brutal workaround.
In file pwlib/include/ptbuildopts.h is defined P_DNS 1 I changed it to P_DNS
0 ... after that recompiled pwlib openh323 and chan_h323 ... make install
from asterisk home dir ... and voila ... no more 60 or (120) seconds delays.
I suppose that this approach is quite graceless... because in this way
entire openh323 DNS resolver is disabled... but this is the only way I
managed to get it working
I'm still looking for proper solution of the problem... so any help or
advice will be appreciated
 

Wow, that *is* brutal.  Still, at least you're working for the moment... 
And another data point for the 60 seconds is *always* DNS rule  !   I 
don't have  h.323 installed here, so I'm of limited utility for 
testing.  What I would do, I think,  is to perform an ethereal trace on  
requests to port 53.  This is simple and will tell you whether the 
problem is inside or outside the asterisk machine.

As DNS appears to be working otherwise, I'll have a look at the h.323 
code again if I get the time today, just in case there's something 
obvious going on.

Oh, and would it be possible for you to post your resolv.conf ?
jd

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Re: [Asterisk-Users] VoiPSupply Dot Com

[snacktime]

 Lastly, we do charge for technical support. We are hear to help, but the
 low margins on ATA's etc certainly does not leave us room to give away
 free support. All of you that are ITSP's know exactly what I am talking
 about.
 
 If you order something, and you can't get it to work, you can pay for us
 to make it work for you. If you order the wrong product, then that is your
 mistake not ours.

You would think this is just common sense  

Personally if I buy something like an ATA, The only thing I expect is
for the product to be as advertised, not to be defective, and to
include adequate documentation to make it work.  And the more bleeding
edge the product, the less 'adequate'  I expect the documentation to
be.

Chris
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Re: [Asterisk-Users] VoipSupply.com

[Danny Froberg]

B. ffs! 

/Danny

On Tue, 2005-05-17 at 22:39 -0500, Brian Capouch wrote:
 Chris Mason wrote:
  I have gotten
  What language is that?
  
 
 Found in an English dictionary:
 
 get
 v. got, (gt) gotten, (gtn)
 v. tr.
 
 You don't like the rules?
 
 B.

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[Asterisk-Users] Iaxtel

[Anton Krall]

Is iaxtel down? Im trying to dial  Echo test: 1700613 and I get a busy
signal... 

Also, is the gw to FWD users down too?

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[Asterisk-Users] Echo Problems

[Terry Wade]

Hi Guys 



I have installed an * system and we seem to have loads of
echo problems. Sometimes worst than others. I have googled and voip-info 
ed my little mind out. I am running 3 x zaphfc cards in the machine. Not sharing
irqs, other than themselves. It is on a PIII 1Gig machine with 1Gb ram. 



My question is this. Does the 2.6 kernel affect (or can) the
echo? 

Could a busy network cause this problem? 

Vmstat  shows cpu usage spike to about 48%



Starting to pull my hair out. Any suggestions would be muchly
appreciated. 



Kind Regards 



Terry Wade

Mobile: +27 82 802-5750

Office: +27 11
784-7642

Fax: +27 11
388-0855



Linux is
like a Wigwam - No gates, no windows, Apache inside



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[Asterisk-Users] callgroup and callwaiting for IAX clients

[IT-PO]

Hi Gurus.
I searched the lists, wiki and the rest of the web but I still do not
understand this.
My Setup is as follows:
[ISDN via chan_capi or IAX2 DiD Provider] = [* PBX] = [IAX2 Clients
(Atcom AT-320ED)]
I want to get callgroup/pickupgroup and callwaiting working on the IAX
phones. Some web sources told me that this was not implemented, others
say that the phone has to handle this, as IAX2 provides these functions
natively.
But how do I get this running? I tried to put callgroup=, pickupgroup=
and callwaiting= statements in my iax.conf, but it does not work.
What do I have to do or how can I workaround if it's really my phones
that have to do this?
--
Best Regards,
Met vriendelijke groeten,
Mit freundlichen Grüßen,
Timm Gebhart
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[Asterisk-Users] find free e1 channel

[borbely . adam]

hi list,
how can i organize several pcs installed with asterisk and e1 cards to be 
seen from an asterisk server as one? so if there is a voip call that needs 
to be forwarded towards the pstn the asterisk server should find a pc that 
has free channel on it's e1 cards that is connected to the pstn side.

/-- Pc1 - E1-\
VOIP - Asterisk--- Pc2 - E1--PBX-PSTN
\-- Pc3 - E1-/
i can not put all e1 cards into one pc since the codec translation that is 
pretty cpu critical.

thank you,
bdz
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[Asterisk-Users] Asterisk with modem

[ALIF Mohssine]

Hi,
Please could any one tell me how could I configure Asterisk inorder to be able to use my modem (instead of FXO cards ...) for outgoing calls. And which type of modems work with Asterisk ?
Do I have to do some changes on Asterisk's scripts or, maybe, add some ones ?!
Thanks in advance.
		 
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Re: [Asterisk-Users] Asterisk with modem

[Dave Cotton]

On Wed, 2005-05-18 at 11:21 +0200, ALIF Mohssine wrote:
 Hi,
 Please could any one tell me how could I configure Asterisk inorder to
 be able to use my modem (instead of FXO cards ...) for outgoing calls.

The simple answer is you can not.

  And which type of modems work with Asterisk ?

None
 
 Do I have to do some changes on Asterisk's scripts or, maybe, add some
 ones ?!

See above

-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Hi,

I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.

The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e

When I'm doing the insmod on zaptel, zaphfc, zaprtc:

Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo 
0xc2d58000(0x2d58000) IRQ 12 HZ 100
zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e

I'm using zaprtc as the gateway is running on a VIA motherboard without USB 
controller.

When I'm doing ztcfg -vv:

Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)

3 channels configured.

Here are my confs:

/etc/zaptel.conf:

loadzone=fr
defaultzone=fr

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

/etc/asterisk/zapata.conf:

[channels]

language=fr
context=test
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
immediate=yes
channel = 1-2

/etc/asterisk/modules.conf:

[modules]
autoload=yes

noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so

noload = app_intercom.so

load = chan_modem.so
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so

noload = chan_alsa.so
noload = chan_oss.so

[global]
chan_modem.so=yes
chan_zap.so=yes


The problem is that after ztcfg ran, I've got the following logs:

Registered tone zone 2 (France)
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 layer 1 state = F3
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623, wanted 8 
got 7), probably a buffer overrun.
zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156, wanted 8 
got 7), probably a buffer overrun.
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes

And when I start asterisk -c, same logs keep on, and I've finally a kernel 
crash:

Unable to handle kernel paging request at virtual address fffc
 printing eip:
 c0113cc0
 *pde = d063
 *pte = 
 Oops: 
 CPU:0
 EIP:0010:[c0113cc0]Not tainted
 EFLAGS: 00010013
 eax: c248015c   ebx:    ecx: 0001   edx: 0001
 esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
 ds: 0018   es: 0018   ss: 0018
 Process sshd (pid: 146, stackpage=c2c8f000)
 Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248 c3819545
0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4 0086
c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008 c270c800
Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7] [c383cd78]
  [c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd] [c0109f78]
  [c010c328]

Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
 0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing

Here is the output from asterisk:

No entry for terminal type screen;
using dumb terminal settings.
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxStatus
  == Manager registered action MailboxCount
  == Manager registered action DBget
  == Manager registered action DBput
  == Manager registered action DBdel
  == Manager registered action ListCommands
  == Parsing '/etc/asterisk/manager.conf': Found
Asterisk Management interface listening on port 5038
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 1 

[Asterisk-Users] Call forward...

[Mark Benson]

Hi,
I'm trying to setup a call forwarding rule so that when an extention 
doesn't answer the call is forwarded to my mobile.

I'm using voiptalk.org for incoming and outgoing calls and SIP phones 
for extentions (so all IP based - no real phone lines).

I tried this (from voip-info.org wiki)...
exten = 1234,1,dial(sip/1234,20)
exten = 1234,2,playback(pls-wait-connect-call)
exten = 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 1234,4,SetCIDNum(0${CALLERIDNUM})
exten = 1234,5,dial(${TRUNK}c/9871234321,20,r)
exten = 1234,6,SetCIDNum(${NewCaller})
exten = 1234,7,voicemail2([EMAIL PROTECTED])
exten = 1234,101,voicemail2([EMAIL PROTECTED])
exten = 1234,102,hangup
Mine looks like this...
exten = 08700688nnn,1,Dial(SIP/operator,1,t)
exten = 08700688nnn,2,playback(pls-wait-connect-call)
exten = 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten = 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten = 08700688nnn,6,SetCIDNum(${NewCaller})
exten = 08700688nnn,7,Voicemail(u100)
exten = 08700688nnn,8,Hangup()
exten = 08700688nnn,101,Voicemail(b100)
exten = 08700688nnn,102,Hangup()
(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but 
that isn't an issue for the time being...
The IAX2/0870n is the extention/device that calls go out on via 
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...

   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, 
c/07961106nnn|20|r) in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to 
create channel of type 'c' (cause 66)

Asterisk shows this from the moment the sip channel is considered not to 
have answered (1 sec)...

   -- Nobody picked up in 1000 ms
   -- Executing Playback(IAX2/[EMAIL PROTECTED]:4569-1, 
pls-wait-connect-call) in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File 
pls-wait-connect-call does not exist in any format
May 18 10:20:26 WARNING[24416]: file.c:790 ast_streamfile: Unable to 
open pls-wait-connect-call (format ilbc): No such file or directory
May 18 10:20:26 WARNING[24416]: app_playback.c:83 playback_exec: 
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569-1 for 
pls-wait-connect-call
   -- Executing SetVar(IAX2/[EMAIL PROTECTED]:4569-1, 
NewCaller=01202843nnn) in new stack
   -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1, 
001202843nnn) in new stack
   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, 
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to 
create channel of type '' (cause 66)
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1, 
01202843nnn) in new stack
   -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]:4569-1, 
u100) in new stack
   -- Playing '/var/spool/asterisk/voicemail/default/100/unavail' 
(language 'en')

Again - I'm not worried about the audio file warning - I can fix that 
later... I guess this is the important bit...

   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, 
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to 
create channel of type '' (cause 66)
 == Everyone is busy/congested at this time (1:0/0/1)

The call then drops into voicemail...
I've tried various permuations but still no call is made to the mobile 
number. Any ideas?

Cheers,
Mark
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Re: [Asterisk-Users] Asterisk with modem

[ALIF Mohssine]

Hello Dave,
Could I know why please ?? Thanks !Dave Cotton [EMAIL PROTECTED] a écrit:
On Wed, 2005-05-18 at 11:21 +0200, ALIF Mohssine wrote: Hi, Please could any one tell me how could I configure Asterisk inorder to be able to use my modem (instead of FXO cards ...) for outgoing calls.The simple answer is you can not. And which type of modems work with Asterisk ?None Do I have to do some changes on Asterisk's scripts or, maybe, add some ones ?!See above-- Dave Cotton <[EMAIL PROTECTED]>___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
		 
Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos !Créez votre Yahoo! Mail 
 
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Re: [Asterisk-Users] Call forward...

[Mark Benson]

I should mention that I have tried using the call forward function of 
the sip phones, but a) this means configuring the phones and some are 
remote and behind firewalls and b) It doesn't work...

Cheers,
Mark

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Re: [Asterisk-Users] zaphfc troubles

[sjaak imap]

Dear Nicolas Olivier
Just try the florz patch at http://zaphfc.florz.dyndns.org/
and look at cat /proc/interupts if your not sharing irq's
Maybe this will help
Good luck
Sjaak
Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing the insmod on zaptel, zaphfc, zaprtc:
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo 
0xc2d58000(0x2d58000) IRQ 12 HZ 100
zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e
I'm using zaprtc as the gateway is running on a VIA motherboard without USB 
controller.
When I'm doing ztcfg -vv:
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
3 channels configured.
Here are my confs:
/etc/zaptel.conf:
loadzone=fr
defaultzone=fr
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
/etc/asterisk/zapata.conf:
[channels]
language=fr
context=test
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
immediate=yes
channel = 1-2
/etc/asterisk/modules.conf:
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so
noload = chan_alsa.so
noload = chan_oss.so
[global]
chan_modem.so=yes
chan_zap.so=yes
The problem is that after ztcfg ran, I've got the following logs:
Registered tone zone 2 (France)
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 layer 1 state = F3
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623, wanted 8 
got 7), probably a buffer overrun.
zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156, wanted 8 
got 7), probably a buffer overrun.
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
And when I start asterisk -c, same logs keep on, and I've finally a kernel 
crash:
Unable to handle kernel paging request at virtual address fffc
printing eip:
c0113cc0
*pde = d063
*pte = 
Oops: 
CPU:0
EIP:0010:[c0113cc0]Not tainted
EFLAGS: 00010013
eax: c248015c   ebx:    ecx: 0001   edx: 0001
esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
ds: 0018   es: 0018   ss: 0018
Process sshd (pid: 146, stackpage=c2c8f000)
Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248 c3819545
   0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4 0086
   c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008 c270c800
Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7] [c383cd78]
 [c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd] [c0109f78]
 [c010c328]
Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing
Here is the output from asterisk:
No entry for terminal type screen;
using dumb terminal settings.
 == Parsing '/etc/asterisk/asterisk.conf': Found
 == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
 == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
 == Manager registered action Ping
 == Manager registered action Events
 == Manager registered action Logoff
 == Manager registered action Hangup
 == Manager registered action Status
 == Manager registered action Setvar
 == Manager registered action Getvar
 == Manager registered action Redirect
 == Manager registered action Originate
 == Manager registered action Command
 == Manager registered action ExtensionState
 == Manager registered action AbsoluteTimeout
 == Manager registered action MailboxStatus
 == Manager registered action MailboxCount
 == Manager registered action DBget
 == Manager registered action DBput
 == Manager registered action DBdel
 == Manager registered action ListCommands
 == Parsing '/etc/asterisk/manager.conf': Found
Asterisk Management interface 

Re: [Asterisk-Users] zaphfc troubles

[Stuart Hirst]

I recently experienced weird buffer overrun errors with zaphfc which I 
eventually identified as being was caused by mismatched memory on the 
motherboard.

You might want to check this out.
Stuart
Nicolas Olivier wrote:
Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing the insmod on zaptel, zaphfc, zaprtc:
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo 
0xc2d58000(0x2d58000) IRQ 12 HZ 100
zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e
I'm using zaprtc as the gateway is running on a VIA motherboard without USB 
controller.
When I'm doing ztcfg -vv:
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
3 channels configured.
Here are my confs:
/etc/zaptel.conf:
loadzone=fr
defaultzone=fr
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
/etc/asterisk/zapata.conf:
[channels]
language=fr
context=test
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
immediate=yes
channel = 1-2
/etc/asterisk/modules.conf:
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so
noload = chan_alsa.so
noload = chan_oss.so
[global]
chan_modem.so=yes
chan_zap.so=yes
The problem is that after ztcfg ran, I've got the following logs:
Registered tone zone 2 (France)
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 layer 1 state = F3
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623, wanted 8 
got 7), probably a buffer overrun.
zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156, wanted 8 
got 7), probably a buffer overrun.
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
And when I start asterisk -c, same logs keep on, and I've finally a kernel 
crash:
Unable to handle kernel paging request at virtual address fffc
printing eip:
c0113cc0
*pde = d063
*pte = 
Oops: 
CPU:0
EIP:0010:[c0113cc0]Not tainted
EFLAGS: 00010013
eax: c248015c   ebx:    ecx: 0001   edx: 0001
esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
ds: 0018   es: 0018   ss: 0018
Process sshd (pid: 146, stackpage=c2c8f000)
Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248 c3819545
   0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4 0086
   c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008 c270c800
Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7] [c383cd78]
 [c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd] [c0109f78]
 [c010c328]
Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing
Here is the output from asterisk:
No entry for terminal type screen;
using dumb terminal settings.
 == Parsing '/etc/asterisk/asterisk.conf': Found
 == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
 == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
 == Manager registered action Ping
 == Manager registered action Events
 == Manager registered action Logoff
 == Manager registered action Hangup
 == Manager registered action Status
 == Manager registered action Setvar
 == Manager registered action Getvar
 == Manager registered action Redirect
 == Manager registered action Originate
 == Manager registered action Command
 == Manager registered action ExtensionState
 == Manager registered action AbsoluteTimeout
 == Manager registered action MailboxStatus
 == Manager registered action MailboxCount
 == Manager registered action DBget
 == Manager registered action DBput
 == Manager registered action DBdel
 == Manager registered action ListCommands
 == Parsing '/etc/asterisk/manager.conf': 

Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Just an update, I deoopsed the kernel dump, must be usable...

Nicolas Olivier wrote:
 
 Hi,
 
 I'm trying to setup a small BRI ISDN - voip gateway.
 The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
 
 The versions i'm running:
 kernel-2.4.27
 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
 zaptel modules 1.0.7
 zaphfc is from bristuff-0.2.0-RC8e
 
 When I'm doing the insmod on zaptel, zaphfc, zaprtc:
 
 Zapata Telephony Interface Registered on major 196
 PCI: Found IRQ 12 for device 00:12.0
 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
 0xc2d58000(0x2d58000) IRQ 12 HZ 100
 zaphfc: Card 0 configured for TE mode
 Registered Span 1 ('ZTHFC1') with 3 channels
 Span ('ZTHFC1') is new master
 zaphfc: 1 hfc-pci card(s) in this box.
 Registered Span 2 ('ZTRTC/1') with 0 channels
 Real Time Clock Driver v1.10e
 
 I'm using zaprtc as the gateway is running on a VIA motherboard without
 USB controller.
 
 When I'm doing ztcfg -vv:
 
 Zaptel Configuration
 ==
 
 SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
 
 Channel map:
 
 Channel 01: Individual Clear channel (Default) (Slaves: 01)
 Channel 02: Individual Clear channel (Default) (Slaves: 02)
 Channel 03: D-channel (Default) (Slaves: 03)
 
 3 channels configured.
 
 Here are my confs:
 
 /etc/zaptel.conf:
 
 loadzone=fr
 defaultzone=fr
 
 span=1,1,3,ccs,ami
 bchan=1-2
 dchan=3
 
 /etc/asterisk/zapata.conf:
 
 [channels]
 
 language=fr
 context=test
 switchtype=euroisdn
 signalling=bri_cpe
 echocancel=yes
 immediate=yes
 channel = 1-2
 
 /etc/asterisk/modules.conf:
 
 [modules]
 autoload=yes
 
 noload = pbx_gtkconsole.so
 noload = pbx_kdeconsole.so
 
 noload = app_intercom.so
 
 load = chan_modem.so
 load = res_features.so
 load = res_musiconhold.so
 load = chan_zap.so
 
 noload = chan_alsa.so
 noload = chan_oss.so
 
 [global]
 chan_modem.so=yes
 chan_zap.so=yes
 
 
 The problem is that after ztcfg ran, I've got the following logs:
 
 Registered tone zone 2 (France)
 zaphfc: card 0 layer 1 state = F4
 zaphfc: card 0 layer 1 state = F5
 zaphfc: card 0 layer 1 state = F7
 zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
 zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
 zaphfc: card 0 layer 1 state = F3
 zaphfc: card 0 layer 1 state = F4
 zaphfc: card 0 layer 1 state = F5
 zaphfc: card 0 layer 1 state = F7
 zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
 wanted 8 got 7), probably a buffer overrun.
 zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
 wanted 8 got 7), probably a buffer overrun.
 zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
 zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
 
 And when I start asterisk -c, same logs keep on, and I've finally a
 kernel crash:
 
 Unable to handle kernel paging request at virtual address fffc
  printing eip:
  c0113cc0
  *pde = d063
  *pte = 
  Oops: 
  CPU:0
  EIP:0010:[c0113cc0]Not tainted
  EFLAGS: 00010013
  eax: c248015c   ebx:    ecx: 0001   edx: 0001
  esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
  ds: 0018   es: 0018   ss: 0018
  Process sshd (pid: 146, stackpage=c2c8f000)
  Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
 c3819545
 0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4
 0086
 c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008
 c270c800
 Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7]
 [c383cd78]
   [c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd]
 [c0109f78]
   [c010c328]
 

EIP; c0113cc0 __wake_up+20/a0   =

eax; c248015c _end+217f0d0/34fef74
esi; c24803a0 _end+217f314/34fef74
edi; c248015c _end+217f0d0/34fef74
ebp; c2c8fe2c _end+298eda0/34fef74
esp; c2c8fe14 _end+298ed88/34fef74

Trace; c3819545 [zaptel]__zt_receive_chunk+133d/1484
Trace; c01cb6b1 __ide_do_rw_disk+3e1/650
Trace; c381aae6 [zaptel]zt_receive+a26/b0c
Trace; c381aad7 [zaptel]zt_receive+a17/b0c
Trace; c383cd78 [zaphfc]hfc_interrupt+228/358
Trace; c01cae16 read_intr+76/1b0
Trace; c383ce95 [zaphfc]hfc_interrupt+345/358
Trace; c01c5416 ide_intr+96/100
Trace; c01cad01 lba_capacity_is_ok+81/120
Trace; c0109ddd handle_IRQ_event+3d/70
Trace; c0109f78 do_IRQ+68/a0
Trace; c010c328 call_do_IRQ+5/d

Code;  c0113cc0 __wake_up+20/a0
 _EIP:
Code;  c0113cc0 __wake_up+20/a0   =
   0:   8b 4b fc  mov0xfffc(%ebx),%ecx   =
Code;  c0113cc3 __wake_up+23/a0
   3:   8b 01 mov(%ecx),%eax
Code;  c0113cc5 __wake_up+25/a0
   5:   85 45 f0  test   %eax,0xfff0(%ebp)
Code;  c0113cc8 __wake_up+28/a0
   8:   74 56 je 60 _EIP+0x60 c0113d20 
__wake_up+80/a0
Code;  c0113cca __wake_up+2a/a0
   a:   31 c0 xor%eax,%eax
Code;  c0113ccc __wake_up+2c/a0
   c:   9cpushf
Code;  c0113ccd __wake_up+2d/a0
   

[Asterisk-Users] eicon fdc3

[Altus Snyman]

Good day all
Did anyone get the eicon 4 bri working with asterisk and fedora core 3
Please
Thanks
Altus

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Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded after 
ztcfg with:

May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun: 
0, 0
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow: 
311, 311
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow: 
436, 436
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun: 
0, 0

And when I start asterisk, same stuff, kernel crashes.

Interrupts are ok.

sjaak imap wrote:
 Dear Nicolas Olivier
 
 Just try the florz patch at http://zaphfc.florz.dyndns.org/
 and look at cat /proc/interupts if your not sharing irq's
 
 Maybe this will help
 
 
 Good luck
 
 Sjaak
 
Hi,

I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.

The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e

When I'm doing the insmod on zaptel, zaphfc, zaprtc:

Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
 0xc2d58000(0x2d58000) IRQ 12 HZ 100
zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e

I'm using zaprtc as the gateway is running on a VIA motherboard without
 USB controller.

When I'm doing ztcfg -vv:

Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)

3 channels configured.

Here are my confs:

/etc/zaptel.conf:

loadzone=fr
defaultzone=fr

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

/etc/asterisk/zapata.conf:

[channels]

language=fr
context=test
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
immediate=yes
channel = 1-2

/etc/asterisk/modules.conf:

[modules]
autoload=yes

noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so

noload = app_intercom.so

load = chan_modem.so
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so

noload = chan_alsa.so
noload = chan_oss.so

[global]
chan_modem.so=yes
chan_zap.so=yes


The problem is that after ztcfg ran, I've got the following logs:

Registered tone zone 2 (France)
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 layer 1 state = F3
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
 wanted 8 got 7), probably a buffer overrun.
zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
 wanted 8 got 7), probably a buffer overrun.
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes

And when I start asterisk -c, same logs keep on, and I've finally a
 kernel crash:

Unable to handle kernel paging request at virtual address fffc
 printing eip:
 c0113cc0
 *pde = d063
 *pte = 
 Oops: 
 CPU:0
 EIP:0010:[c0113cc0]Not tainted
 EFLAGS: 00010013
 eax: c248015c   ebx:    ecx: 0001   edx: 0001
 esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
 ds: 0018   es: 0018   ss: 0018
 Process sshd (pid: 146, stackpage=c2c8f000)
 Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
 c3819545
0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4
 0086
c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008
 c270c800
Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7]
 [c383cd78]
  [c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd]
 [c0109f78]
  [c010c328]

Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
 0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing

Here is the output from asterisk:

No entry for terminal type screen;
using dumb terminal settings.
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered 

[Asterisk-Users] Call forwarding...

[Mark Benson]

Sorry for posting this again, but it seems to have become attached to 
another thread. Guess I replied to another message instead of starting a 
new one...

Hi,
I'm trying to setup a call forwarding rule so that when an extention 
doesn't answer the call is forwarded to my mobile.

I'm using voiptalk.org for incoming and outgoing calls and SIP phones 
for extentions (so all IP based - no real phone lines).

I tried this (from voip-info.org wiki)...
exten = 1234,1,dial(sip/1234,20)
exten = 1234,2,playback(pls-wait-connect-call)
exten = 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 1234,4,SetCIDNum(0${CALLERIDNUM})
exten = 1234,5,dial(${TRUNK}c/9871234321,20,r)
exten = 1234,6,SetCIDNum(${NewCaller})
exten = 1234,7,voicemail2([EMAIL PROTECTED])
exten = 1234,101,voicemail2([EMAIL PROTECTED])
exten = 1234,102,hangup
Mine looks like this...
exten = 08700688nnn,1,Dial(SIP/operator,1,t)
exten = 08700688nnn,2,playback(pls-wait-connect-call)
exten = 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten = 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten = 08700688nnn,6,SetCIDNum(${NewCaller})
exten = 08700688nnn,7,Voicemail(u100)
exten = 08700688nnn,8,Hangup()
exten = 08700688nnn,101,Voicemail(b100)
exten = 08700688nnn,102,Hangup()
(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but 
that isn't an issue for the time being...
The IAX2/0870n is the extention/device that calls go out on via 
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...

  -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, 
c/07961106nnn|20|r) in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to 
create channel of type 'c' (cause 66)

Asterisk shows this from the moment the sip channel is considered not to 
have answered (1 sec)...

  -- Nobody picked up in 1000 ms
  -- Executing Playback(IAX2/[EMAIL PROTECTED]:4569-1, 
pls-wait-connect-call) in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File 
pls-wait-connect-call does not exist in any format
May 18 10:20:26 WARNING[24416]: file.c:790 ast_streamfile: Unable to 
open pls-wait-connect-call (format ilbc): No such file or directory
May 18 10:20:26 WARNING[24416]: app_playback.c:83 playback_exec: 
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569-1 for 
pls-wait-connect-call
  -- Executing SetVar(IAX2/[EMAIL PROTECTED]:4569-1, 
NewCaller=01202843nnn) in new stack
  -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1, 
001202843nnn) in new stack
  -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, 
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to 
create channel of type '' (cause 66)
== Everyone is busy/congested at this time (1:0/0/1)
  -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1, 
01202843nnn) in new stack
  -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]:4569-1, 
u100) in new stack
  -- Playing '/var/spool/asterisk/voicemail/default/100/unavail' 
(language 'en')

Again - I'm not worried about the audio file warning - I can fix that 
later... I guess this is the important bit...

  -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, 
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to 
create channel of type '' (cause 66)
== Everyone is busy/congested at this time (1:0/0/1)

The call then drops into voicemail...
I've tried various permuations but still no call is made to the mobile 
number. Any ideas?

Cheers,
Mark
I should mention that I have tried using the call forward function of 
the sip phones, but a) this means configuring the phones and some are 
remote and behind firewalls and b) It doesn't work...

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Re: [Asterisk-Users] Re: cisco 3620 setup (newbie cisco alert)

[Asterisk]

Arrggh Nuts.
Don't suppose anyone has a spare NM-HDV hanging around that they want to 
sell ?

:(
Julian.
[EMAIL PROTECTED] wrote:
You need an NM-HDV card of some sort to run voice.  The WIC-1MFT-E1 
can handle voice, but you still need the DSP's to use it as a voice 
card. Putting that into an NM-HDV that has DSP's will make the voice 
ports and dsp's show up.

Asterisk wrote:
ok, I'm starting to get confused, you must be getting annoyed .. how 
do I add this voice-port ?

Also, if I want to use * to handle all of the SIP internal calls, and 
cisco to handle to inbound and outbound isdn-32 PRI calls, what 
feature set of the IOS should I need ? What hardware on the cisco 
3620 should I need ? I've got a PRI-1CE1U and a WIC-1MFT-E1 and a 
two-port fast ethernet. I also have a 2610 to play with if required :)

Many thanks for your help. It is much appreciated. There is only so 
much a newbie can do without advice.

Julian.
barney wrote:
I can`t see voice-port in your configuration. Something like this:
!
voice-port 1/0:15; voice-port 1/0:15, if your D channel 
is Serial1/0:15
input gain -6
output attenuation 14
echo-cancel coverage 32
echo-cancel suppressor
cptone SK
description E1
bearer-cap Speech
!

If you configure voice-port, try again to configure dial-peer (port 
x/y:15 command) and if you have DSPs in you box, you must see 
something like this:

SIP-3640#show voice dsp
DSP  DSP DSPWARE CURR  BOOT PAK 
TX/RX
TYPE NUM CH CODECVERSION STATE STATE   RST AI VOICEPORT TS 
ABORT  PACK COUNT
 === ==  === = === === == = == = 

C549 000 00 {high}3.4.56 IDLE  idle  0  0 2/0:1501 0 
0/16
01 {high}3.4.56 IDLE  idle 0 2/0:1517 0 0/0
C549 001 00 {high}3.4.56 IDLE  idle  0  0 2/0:1502 0 
0/16
01 {high}3.4.56 IDLE  idle 0 2/0:1518 0 0/0


If you don`t have any DSP in your C3620, it will be unusable. So 
check it!

-b
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Re: [Asterisk-Users] Call forwarding...

[Peter Bowyer]

In 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
 
 I'm trying to setup a call forwarding rule so that when an extention
 doesn't answer the call is forwarded to my mobile.
 
 I'm using voiptalk.org for incoming and outgoing calls and SIP phones
 for extentions (so all IP based - no real phone lines).
 
 I tried this (from voip-info.org wiki)...
 
 exten = 1234,1,dial(sip/1234,20)
 exten = 1234,2,playback(pls-wait-connect-call)
 exten = 1234,3,Setvar(NewCaller=${CALLERIDNUM})
 exten = 1234,4,SetCIDNum(0${CALLERIDNUM})
 exten = 1234,5,dial(${TRUNK}c/9871234321,20,r)
 exten = 1234,6,SetCIDNum(${NewCaller})
 exten = 1234,7,voicemail2([EMAIL PROTECTED])
 exten = 1234,101,voicemail2([EMAIL PROTECTED])
 exten = 1234,102,hangup
 
 Mine looks like this...
 
 exten = 08700688nnn,1,Dial(SIP/operator,1,t)
 exten = 08700688nnn,2,playback(pls-wait-connect-call)
 exten = 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
 exten = 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
 exten = 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
 exten = 08700688nnn,6,SetCIDNum(${NewCaller})
 exten = 08700688nnn,7,Voicemail(u100)
 exten = 08700688nnn,8,Hangup()
 exten = 08700688nnn,101,Voicemail(b100)
 exten = 08700688nnn,102,Hangup()
 
 (where nnn is a real number)
 The sip channel is set to time out quickly for testing.
 And I don't appear to have the pls-wait-connect-call audio file - but
 that isn't an issue for the time being...
 The IAX2/0870n is the extention/device that calls go out on via
 voiptalk... (my call provider)...
 If I include the c/ in the TRUNK line I get...
 
   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
 c/07961106nnn|20|r) in new stack
 May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
 type registered for 'c'
 May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
 create channel of type 'c' (cause 66)

Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] DHCP, PoE, FXS, FXO and ONE power adapter ONLY???

[Ronald Wiplinger]

This afternoon we were discussing, and found that we would like one box, 
which should have ALL of these:

1. WAN port
2. Ethernet port 1 with Power over Ethernet
3. Ethernet port 2 with or without PoE
4. FXS port
5. FXO port
6. DHCP, web configureable.
7. Optional wireless accesspoint
8. One and ONLY one power adapter for this box
Does such a box exist?
 

bye
Ronald
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[Asterisk-Users] HELP ME!!!! Asterisk don't do calls

[Michele \O-Zone\ Pinassi]

Hi all,

as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions:

moloch*CLI sip show peers
Name/usernameHostDyn Nat ACL Mask Port Status
204/204  (Unspecified)D  255.255.255.255  0UNKNOWN
203/203  192.167.125.9D  255.255.255.255  5062 OK (3 ms)
202/202  (Unspecified)D  255.255.255.255  0UNKNOWN
201/201  192.167.125.12   D  255.255.255.255  5060 OK (3 ms)
moloch*CLI

as you can see, 201 and 203 are on-line but, if i call from 203 to 201, i immediately go to voicemail instead of doing call to 201. Here's the SIP call flow:

moloch*CLI

Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK64FE3538
CSeq: 1114 INVITE
To: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
From: 203 sip:[EMAIL PROTECTED];tag=1CE28F8
Call-ID: [EMAIL PROTECTED]
Subject: sip:[EMAIL PROTECTED]
Content-Length: 187
User-Agent: kphone/4.0.5
Contact: 203 sip:[EMAIL PROTECTED]:5062;transport=udp

v=0
o=username 0 0 IN IP4 192.167.125.9
s=The Funky Flow
c=IN IP4 192.167.125.9
t=0 0
m=audio 36808 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

11 headers, 9 lines
Using latest request as basis request
Sending to 192.167.125.9 : 5062 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK64FE3538
From: 203 sip:[EMAIL PROTECTED];tag=1CE28F8
To: sip:[EMAIL PROTECTED];tag=as3c1a1273
Call-ID: [EMAIL PROTECTED]
CSeq: 1114 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=0ae53906
Content-Length: 0


 to 192.167.125.9:5062
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Found user '203'
moloch*CLI

Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK64FE3538
CSeq: 1114 ACK
To: sip:[EMAIL PROTECTED];tag=as3c1a1273
From: 203 sip:[EMAIL PROTECTED];tag=1CE28F8
Call-ID: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.0.5
Contact: 203 sip:[EMAIL PROTECTED]:5062;transport=udp


9 headers, 0 lines
moloch*CLI

Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK7602DA72
CSeq: 1115 INVITE
To: sip:[EMAIL PROTECTED]
Proxy-Authorization: Digest username=203, realm=asterisk, nonce=0ae53906, uri=sip:[EMAIL PROTECTED], cnonce=abcdefghi, nc=0001, response=58e82c67b3c712ffb39220e473903007, opaque=, algorithm=MD5
Content-Type: application/sdp
From: 203 sip:[EMAIL PROTECTED];tag=1CE28F8
Call-ID: [EMAIL PROTECTED]
Subject: sip:[EMAIL PROTECTED]
Content-Length: 187
User-Agent: kphone/4.0.5
Contact: 203 sip:[EMAIL PROTECTED]:5062;transport=udp

v=0
o=username 0 0 IN IP4 192.167.125.9
s=The Funky Flow
c=IN IP4 192.167.125.9
t=0 0
m=audio 36808 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000

12 headers, 9 lines
Using latest request as basis request
Sending to 192.167.125.9 : 5062 (non-NAT)
Found user '203'
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Peer audio RTP is at port 192.167.125.9:36808
Found description format PCMU
Found description format GSM
Found description format iLBC
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x406 (gsm|ulaw|ilbc)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 201 in from-internal
list_route: hop: sip:[EMAIL PROTECTED]:5062;transport=udp
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK7602DA72
From: 203 sip:[EMAIL PROTECTED];tag=1CE28F8
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 1115 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


 to 192.167.125.9:5062
-- Executing Macro(SIP/203-f9ee, exten-vm|[EMAIL PROTECTED]|201) in new stack
-- Executing SetVar(SIP/203-f9ee, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/203-f9ee, 0?novm|1:3) in new stack
-- Goto (macro-exten-vm,s,3)
-- Executing GotoIf(SIP/203-f9ee, 0?novm|1) in new stack
-- Executing Macro(SIP/203-f9ee, dial|30|tr|201) in new stack
-- Executing AGI(SIP/203-f9ee, dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- AGI Script dialparties.agi completed, returning 0
-- Executing Wait(SIP/203-f9ee, 1) in new stack
-- Executing VoiceMail(SIP/203-f9ee, [EMAIL PROTECTED]) in new stack
We're at 192.167.125.9 port 15724
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x2 (gsm)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 

[Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)

[Lee Norvall]

Hi
 
I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server.  We can
use all 4 lines for out going calls fine, but on incoming we can only use 2.
On calling in using the main msn, the 3rd line gives a an engaged signal.

I have unplugged 1 of the cards, and the other card takes the 2 calls.  I
then swapped this around, and this also works fine.  But when using both
cards, we can only use 2 line in.

Any ideas???

Rgds
 

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RE: [Asterisk-Users] voicemail.conf from DB

[Senad J]

[EMAIL PROTECTED] wrote:
 Hi

 tks for the feedback, the admintool i cant use, because users
 create/add themselves to the system themselves, could be 100 or 1000+
 users. Hence I could get my script which create user/pass details in
 myqsql to call the voicemail script to create the physical path on
 the server, but appending 1000+ lines to voicemail.conf doesnt seem
 like a good idea, and then reloading each time.
 I looked at the realtime feature in asterisk and it looked good,


yap... it sounds promissing


 want to store the voicemail on the server itself, (wish i could
 change the dir struture though, rather than have one
 ../context/voicemail etc I would want it split like a mbox mail
 directory structure for large scalabilty---but thats another
 story-new thread),

Why not use GFS or similar for this purpose. Since GFS it is global/cluster
file system
you can expand it, it is fully posix complient etc. and you do not need to
worry about
creating ../context/voicemail.



 and the mapping , user/pass details for each user
 pull from the DB.

 Iqbal

 On 5/16/2005, Senad J [EMAIL PROTECTED] wrote:

 [EMAIL PROTECTED] wrote:
 but the other choice is to keep editing the voicemail.conf file,
 everytime I add a new user, which again is not really scalable.

 Using an administration interface of some kind will solve this issue.

 I dont wish to store the voicemail in the DB, just the conf file
 itself, mysql will easily in a clusered scenario support 100K+
 entries.


 Fine...
 What setup would you recommend

 Do do what?

 Senad
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Re: [Asterisk-Users] DHCP, PoE, FXS, FXO and ONE power adapter ONLY???

[Iqbal]

doesnt invetel do one
Iqbal
Ronald Wiplinger wrote:
This afternoon we were discussing, and found that we would like one 
box, which should have ALL of these:

1. WAN port
2. Ethernet port 1 with Power over Ethernet
3. Ethernet port 2 with or without PoE
4. FXS port
5. FXO port
6. DHCP, web configureable.
7. Optional wireless accesspoint
8. One and ONLY one power adapter for this box
Does such a box exist?

bye

Ronald
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[Asterisk-Users] FWD to Asterisk stops after 3 seconds

[Ronald Wiplinger]

I asked my friend to setup FWD and call me to my *
However, it did not matter which codec we used, after three seconds the 
connection was cut.

Why? and how to make it stabled?
bye
Ronald
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Re: [Asterisk-Users] Call forwarding...

[Mark Benson]

Er... set the trunk variable to what? I thought it was a built in 
variable...

Peter Bowyer wrote:
Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.
Peter
 

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[Asterisk-Users] Asterisk with Intel modems 537 or MD3200

[ALIF Mohssine]

I've just give a look to the website 
http://www.voip-info.org/wiki-Asterisk+Hardware
If I understand very well, the Intel modems marked with 537 or MD3200 chipset should work with Asterisk ?!
If it is true, I'd like to know how to configure Asterisk ?
Thanks a lot.
		 
Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos !Créez votre Yahoo! Mail 
 
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[Asterisk-Users] IVR/Voicemail, No Sound from Asterisk

[Robson Ribeiro]

Hi all, 



I am having a problem with a recent installed *. The IVR,
voicemail internal greeting sounds dont play!. I see on the CLI
interface that it is playing but I cant hear anything.

I have the following configuration on the asterisk.



-
Current Asterisk CVS

-
A TDM400 with 4 FXOs

-
A FRITZ ISDN using CAPI

-
Linux Debian 2.4.27



Thanks.



Robson 








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Re: [Asterisk-Users] Call forwarding...

[Peter Bowyer]

On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
 Er... set the trunk variable to what? I thought it was a built in
 variable...

No, it's not. Looking at your dialplan extract, you need to set TRUNK
to the name of the trunk to place the outgoing call on.

eg

TRUNK=IAX/voiptalk

You might need to mess around to get the dialstring to end up in the
right format for the provider you're using, also. Or imbed it directly
in the dialplan.

Peter
-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] Asterisk H323 Trunk Zone

[Mahmoud Badran]

AVE!

i am trying to register h323 asterisk to the gatekeeper as i installed
asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323
on fedora core3 on a cisco mcs 7800 server problem is i want the
asterisk to register with gatekeeper endpoint with specific zone name
and type...

i searched the web, mail list but there weren't any helpful ones 

could anyone plz tell me how to specify the zone name and type??

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Re: [Asterisk-Users] Call forwarding...

[Mark Benson]

I have been able to get it working by explicitly setting the dial command...
So should the trunk variable be the divice to dial out on?
Mark Benson wrote:
Er... set the trunk variable to what? I thought it was a built in 
variable...

Peter Bowyer wrote:
Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.
Peter
 

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Re: [Asterisk-Users] multiple sip accounts from same sip registrar

[Matt Scott]

Hi Peter.

I think I probably put my email rather badly.
However you did manage to spot my problem and solve it for which I am very
grateful!!

The bottom line is you cannot have different context for the same sip
provider, and it works as you state in your reply.

Thanks again.

Matt
- Original Message -
From: Peter Bowyer [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 18, 2005 8:25 AM
Subject: Re: [Asterisk-Users] multiple sip accounts from same sip registrar


On 17/05/05, Matt Scott [EMAIL PROTECTED] wrote:
 Dear all,

 I have an asterisk sip issue which I don't believe is unique.
 I use a registrar (sipgate.co.uk) where I have 3 different accounts.
 These accounts provide me with three seperate local phone numbers which
 allow me to allocate them to seperate users.
 By using just one of these accounts I can set asterisk up to send and
 receive calls no problem.
 However, when I start to introduce an additional account I start to run
into
 problems.

 if I do a 'sip show peers' with a good config I think it may outline the
 problem

 sip show peers
 Name/username  HostDyn Nat ACL Mask
Port
 Status
 1005/1005  (Unspecified)D  255.255.255.255  0
 Unmonitored
 1004/1004  (Unspecified)D  255.255.255.255  0
 Unmonitored
 1003/1003  (Unspecified)D  255.255.255.255  0
 Unmonitored
 1002/1002  10.0.0.52D  255.255.255.255
5060
 Unmonitored
 1001/1001  10.0.0.51D  255.255.255.255
5060
 Unmonitored
 sipgate1/321   217.10.79.219N  255.255.255.255
5060
 OK (52 ms)

I'm not sure what you think the problem is, you haven't told us... but
anyway, I haven't succeeded in sending sipgate inbound calls through
separate contexts, but I deal with them all in a single context - the
calls will arrive at an extension matching the individual sipgate
username in the register command.

Works for me and several others

Peter


--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] VoiPSupply Dot Com

[Rich Adamson]

  Being around the internet for a quite a long time this gives me an 
  uneasy feeling. I have seen company’s start to go under and pull the 
  plug when they get into financial trouble(not being able to pay the 
  bills) and run with the customers money. I have had this happen to me 
  on 2 occasions. Just the woes of doing business on the net.
 
  Being in Canada it makes it very difficult to find companies that will 
  ship COD from the US. If I was to order I would only order COD from 
  now on from VoipSupply.
 
  I have ordered product from VoipSupply and received the product. I 
  will not be ordering more product do to this outage of the phones with 
  no explanation.
 
  Just my 2cents.
 
 Maybe the tollfree provider was responsible for the outage and maybe it 
 only affected service from Canada.

Or, maybe they were using Broadvoice. ;)


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Re: [Asterisk-Users] zaphfc troubles

[Stuart Hirst]

Nicolas,
I replied earlier stating that I saw similar issues and now that you 
have applied the Florz patch the symptoms you are seeing are all but 
identical to the issues I saw and resolved by changing out the 
motherboard memory. The system was an ASUS main board with a Xeon processor.

It is not the memory it could be something specific to the VIA motherboard.
Stuart

Nicolas Olivier wrote:
Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded after 
ztcfg with:
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun: 
0, 0
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow: 
311, 311
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow: 
436, 436
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun: 
0, 0
And when I start asterisk, same stuff, kernel crashes.
Interrupts are ok.
sjaak imap wrote:
 

Dear Nicolas Olivier
Just try the florz patch at http://zaphfc.florz.dyndns.org/
and look at cat /proc/interupts if your not sharing irq's
Maybe this will help
Good luck
Sjaak
   

Hi,
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing the insmod on zaptel, zaphfc, zaprtc:
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
 

0xc2d58000(0x2d58000) IRQ 12 HZ 100
   

zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e
I'm using zaprtc as the gateway is running on a VIA motherboard without
 

USB controller.
   

When I'm doing ztcfg -vv:
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
3 channels configured.
Here are my confs:
/etc/zaptel.conf:
loadzone=fr
defaultzone=fr
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
/etc/asterisk/zapata.conf:
[channels]
language=fr
context=test
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
immediate=yes
channel = 1-2
/etc/asterisk/modules.conf:
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so
noload = chan_alsa.so
noload = chan_oss.so
[global]
chan_modem.so=yes
chan_zap.so=yes
The problem is that after ztcfg ran, I've got the following logs:
Registered tone zone 2 (France)
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 layer 1 state = F3
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
 

wanted 8 got 7), probably a buffer overrun.
   

zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
 

wanted 8 got 7), probably a buffer overrun.
   

zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
And when I start asterisk -c, same logs keep on, and I've finally a
 

kernel crash:
   

Unable to handle kernel paging request at virtual address fffc
printing eip:
c0113cc0
*pde = d063
*pte = 
Oops: 
CPU:0
EIP:0010:[c0113cc0]Not tainted
EFLAGS: 00010013
eax: c248015c   ebx:    ecx: 0001   edx: 0001
esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
ds: 0018   es: 0018   ss: 0018
Process sshd (pid: 146, stackpage=c2c8f000)
Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
 

c3819545
   

  0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4
 

0086
   

  c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008
 

c270c800
   

Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7]
 

[c383cd78]
   

[c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd]
 

[c0109f78]
   

[c010c328]
Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing
Here is the output from asterisk:
No entry for terminal type screen;
using dumb terminal settings.
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 

Re: [Asterisk-Users] Asterisk - Spandsp: fax header

[Jean-Yves Avenard]

HelloOn 18/05/2005, at 4:09 PM, Peter Svensson wrote:I think he is refering to the remote fax id to be presented, not the  header. I.e. the 20 digit user selectable number on the remote fax. The  one often seen on the lcd of the receiving fax and so on. Yes that's exactly what I'm referring to.Most fax machines I've used print this information on the top left corner or top right corner on any fax received.Is it possible to do this with SpanDSP?Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200  ___
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[Asterisk-Users] Asterisk and Ericsson PBX

[j.peran.fernandez]

Hi, I´m trying to migrate my propietary software to an asterisk server 
connected to a Ericsson BP 128i PBX.

I´ve been looking at the asterisk web, user forums, published docs about how to 
use the PBX as the hardware device but I haven´t found anything.

I think this is possible. The old server is currently connected to the Ericcson 
via serial port.

Please, help.

Thanks a lot.



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Re: [Asterisk-Users] DHCP, PoE, FXS, FXO and ONE power adapter ONLY???

[Andrew Kohlsmith]

On May 18, 2005 06:45 am, Iqbal wrote:
 doesnt invetel do one

Got a link?  Googling for invetel comes up with car counters and stuff...  
nothing really VOIP related.

-A.
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Re: [Asterisk-Users] Call forward...

[adria vidal]

El 18/05/2005, a las 11:42, Mark Benson escribió:
-- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1, 
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel 
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable 
to create channel of type '' (cause 66)
 == Everyone is busy/congested at this time (1:0/0/1)

The call then drops into voicemail...

Maybe you have to erase the  in your Trunk variable ?

··
Adrià Vidal
...

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Re: [Asterisk-Users] Call forwarding...

[Mark Benson]

Thanks,
Staring to see where I was going wrong. Now I know the explicit dial 
string (as you say I tried that in the dial plan and it worked) I can 
mess around with the trunk variable.

Cheers!
Peter Bowyer wrote:
On 18/05/05, Mark Benson [EMAIL PROTECTED] wrote:
 

Er... set the trunk variable to what? I thought it was a built in
variable...
   

No, it's not. Looking at your dialplan extract, you need to set TRUNK
to the name of the trunk to place the outgoing call on.
eg
TRUNK=IAX/voiptalk
You might need to mess around to get the dialstring to end up in the
right format for the provider you're using, also. Or imbed it directly
in the dialplan.
Peter
 

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[Asterisk-Users] Asterisk not recognising On Hold

[Erik Versaevel - Infopact Netwerkdiensten BV]

I'm having some troubles with my * machine, when i place a call on hold
the callee doesn't hear any MOH and the call is dropped because of lack
of RTP.
I also don't see * starting MOH on the SIP channel the callee is on (moh
class is defined, there are MP3 files and mpg123 is active).

I'm using * 1.0.6 right now with Cisco 7940's, i can see * recieving a
SIP invite with c=0.0.0.0 so that should work, i can allso see the
invite back to the phone when getting the call out of hold. Because of
this problem attented transfers won't work correct either (since the
other side of the call gets dropped before the call is transferred). All
calls are SIP--SIP.

Any ideas?

Kind regards,

E. Versaevel


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Re: [Asterisk-Users] Asterisk - Spandsp: fax header

[Andrew Kohlsmith]

On May 18, 2005 07:22 am, Jean-Yves Avenard wrote:
 Yes that's exactly what I'm referring to.
 Most fax machines I've used print this information on the top left
 corner or top right corner on any fax received.

 Is it possible to do this with SpanDSP?

You can get the info and stamp it into the image yourself with some third 
party TIFF manipulation tools, I bet.

rxfax is a simple fax reception app; if you need more than what it offers you 
you have several options, but they all involve work.  :-)  I think Steve's 
been very clear about what rxfax can and can't do.

-A.
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Re: [Asterisk-Users] zaphfc troubles

[Peer Oliver Schmidt]

Nicolas Olivier wrote:
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing the insmod on zaptel, zaphfc, zaprtc:
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo 
0xc2d58000(0x2d58000) IRQ 12 HZ 100
zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e
I'm using zaprtc as the gateway is running on a VIA motherboard without USB controller.
[..]
Why are you running zaprtc? zaphfc provides your needed timing source.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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Re: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)

[Armin Schindler]

On Wed, 18 May 2005, Lee Norvall wrote:
 Hi
  
 I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server.  We can
 use all 4 lines for out going calls fine, but on incoming we can only use 2.
 On calling in using the main msn, the 3rd line gives a an engaged signal.
 
 I have unplugged 1 of the cards, and the other card takes the 2 calls.  I
 then swapped this around, and this also works fine.  But when using both
 cards, we can only use 2 line in.

There are two possibilities:

1) your Telco doesn't send the 3rd call to your other line.
   You can verify that by using
divactrl mlog -c 1 -o  (diva_idi module must be leaded)
   and see if an incoming call is shown.
   (use -c 2 for the second card)

2) your configuration of chan_capi is not correct and the 3rd call is 
   ignored/rejected.

If you don't use DIVA Server cards with CAPI, forget this mail.

Armin

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Re: [Asterisk-Users] Asterisk - Spandsp: fax header

[Jean-Yves Avenard]

HiOn 18/05/2005, at 9:35 PM, Andrew Kohlsmith wrote:You can get the info and stamp it into the image yourself with some third  party TIFF manipulation tools, I bet. I wouldn't mind doing so if I knew where this Fax ID information is stored or how to retrieve it, or if it's even possible.JY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200  ___
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RE: [Asterisk-Users] fax soft client

[Dean Collins]

Wow looks perfect - this will be unreal if this works.

Dean

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Peter Valkov
 Sent: Wednesday, 18 May 2005 12:14 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] fax soft client
 
 check this http://www.inter7.com/?page=astfax
 
 I'm planning to use it but did not have time to check how it works
 
 --- Dean Collins [EMAIL PROTECTED] wrote:
 
  I emailed this before but never got a reply, maybe there have been
some
  new technical developments.
 
 
 
  I understand that the AMP/[EMAIL PROTECTED] now allows faxes to be received 
  via a
  software solution but I'm interested, is there a way to send faxes
using
  software? Maybe something like a SIP or IAX software client that can
be
  run on a XP pc to initiate the call to an asterisk server?
 
 
 
  Cheers,
 
  Dean
 
 
 
 
 
 
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Re: [Asterisk-Users] Asterisk with modem

[Rich Adamson]

  Please could any one tell me how could I configure Asterisk inorder to
  be able to use my modem (instead of FXO cards ...) for outgoing calls.
 
 The simple answer is you can not.
 
   And which type of modems work with Asterisk ?
 
 None
  
  Do I have to do some changes on Asterisk's scripts or, maybe, add some
  ones ?!
 
 See above

Well, you sort of can use a modem (kind of off topic). One can
configure a modem as a PPP dialup link into an isp, and then
use that link with low data rate codecs to place sip/iax calls.
But, as Dave pointed out, its not a means to substitute for an
analog fxo-pstn interface.


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Re: [Asterisk-Users] DEBUG output on sip extensions

[Mark Johnson]

Marty Mastera wrote:
Can anyone help me to understand what the significance of this output is?
 
May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4
May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels 
SIP/105-1ae4 and SIP/outbound-7dc3
 
I searched for these phrases but am coming up short on what they 
really mean.  I'm trying to investigate problems we are having with 
two separate asterisk installations both using Polycom IP-500 phones.  
These type of messages appear in the logs of both servers.  It almost 
appears as though these messages are normal following completion of a 
call (a hangup), but we are troubleshooting bad audio in both 
locations and the wording of these messages doesn't appear benign.
 
I am noticing these in my logs also.  I looks like it is the result of 
the person hanging up, but I have had a few comlaints of dropped calls 
the last few days.  These messages also appear at the times of the 
dropped calls.  I have been watching CPU usage and it doesn't look like 
my machine was really loaded or anything.

Mark
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Re: [Asterisk-Users] Asterisk - Spandsp: fax header

[Steve Underwood]

Jean-Yves Avenard wrote:
Hello
On 18/05/2005, at 4:09 PM, Peter Svensson wrote:
I think he is refering to the remote fax id to be presented, not the 

header. I.e. the 20 digit user selectable number on the remote fax. The 

one often seen on the lcd of the receiving fax and so on.

Yes that's exactly what I'm referring to.
Most fax machines I've used print this information on the top left 
corner or top right corner on any fax received.

Is it possible to do this with SpanDSP?
Jean-Yves
It is only there because the sending machine put it there in the image. 
Spandsp is not different from how any FAX machine I have ever used 
behaves. As well as sending the 20 digit number as text, the sending 
machine puts in the header.

Regards,
Steve
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[Asterisk-Users] Traffic shaping for IAX and SIP calls through Asterisk?

[Mike Dent]

Hi,
Is it possible to put some kind of bridge which will do traffic
shaping/prioritising between
my 6 external IP addresses and my PPPoA modem interface?
My other option is to put some kind of device at the edge of all my
networks to shape the
traffic in/out. I'd rather do it in one box if possible?

thanks

Mike
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RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)

[Lee Norvall]

Hi

I can see what seems to be both devices in use, so I guess it must be
down to the capi.conf (below), does this look correct ???

[interfaces]

msn=292880
incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886,
292887, 292888, 292889
outgoingmsn=292880
controller=1
softdtmf=1
;accountcode=
context=demo
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

msn=292xxx
incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886,
292887, 292888, 292889
outgoingmsn=292880
controller=2
softdtmf=1
;accountcode=
context=demo
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: 18 May 2005 12:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)

On Wed, 18 May 2005, Lee Norvall wrote:
 Hi
  
 I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server.  We
can
 use all 4 lines for out going calls fine, but on incoming we can only
use 2.
 On calling in using the main msn, the 3rd line gives a an engaged
signal.
 
 I have unplugged 1 of the cards, and the other card takes the 2 calls.
I
 then swapped this around, and this also works fine.  But when using
both
 cards, we can only use 2 line in.

There are two possibilities:

1) your Telco doesn't send the 3rd call to your other line.
   You can verify that by using
divactrl mlog -c 1 -o  (diva_idi module must be leaded)
   and see if an incoming call is shown.
   (use -c 2 for the second card)

2) your configuration of chan_capi is not correct and the 3rd call is 
   ignored/rejected.

If you don't use DIVA Server cards with CAPI, forget this mail.

Armin

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RE: [Asterisk-Users] DEBUG output on sip extensions

[Andreas Sikkema]

[EMAIL PROTECTED] wrote:

 Marty Mastera wrote:
 May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel:
 SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging
 channels SIP/105-1ae4 and SIP/outbound-7dc3

 I am noticing these in my logs also.  I looks like it is the result
 of the person hanging up, but I have had a few comlaints of
 dropped calls
 the last few days.  These messages also appear at the times of the
 dropped calls.  I have been watching CPU usage and it doesn't look
 like my machine was really loaded or anything.

I see these with every single call. I (naturally I'd say) also 
have reports of dropped calls, but have never been able to relate 
them to these messages. The messages happen much more often.

-- 
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544  f: +31 (0)10 413 65 45
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Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Quoting from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation
As I haven't got a Digium card, I need a timer which can be provided by 
ztdummy, zaprtc or zaprai.

But anyway the results are the same with or without zaprtc loaded.

Peer Oliver Schmidt wrote:
 Nicolas Olivier wrote:
 
 I'm trying to setup a small BRI ISDN - voip gateway.
 The ISDN card is based on Cologne chipset, so I try set it up with
 zaphfc.

 The versions i'm running:
 kernel-2.4.27
 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
 zaptel modules 1.0.7
 zaphfc is from bristuff-0.2.0-RC8e

 When I'm doing the insmod on zaptel, zaphfc, zaprtc:

 Zapata Telephony Interface Registered on major 196
 PCI: Found IRQ 12 for device 00:12.0
 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
 0xc2d58000(0x2d58000) IRQ 12 HZ 100
 zaphfc: Card 0 configured for TE mode
 Registered Span 1 ('ZTHFC1') with 3 channels
 Span ('ZTHFC1') is new master
 zaphfc: 1 hfc-pci card(s) in this box.
 Registered Span 2 ('ZTRTC/1') with 0 channels
 Real Time Clock Driver v1.10e

 I'm using zaprtc as the gateway is running on a VIA motherboard
 without USB controller.
 [..]
 
 Why are you running zaprtc? zaphfc provides your needed timing source.
 -- 
 Best regards
 
 Peer Oliver Schmidt
 PGP Key ID: 0x83E1C2EA
 
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[Asterisk-Users] SIP/nat situation

[Pizco Dominguez]

Hi.

We are trying to set up asterisk to service a wireless community in our
town.

We have about 30/40 wireless working nodes each one with a 10.34.x.x/24
subnet for users. Each one of these addresses can potentially have a
192.168.x.x/x subnet.

On top, the wireless nodes, themselves, are linked in 172.16.x.x/x
subnets.

On top of the top, there is internet and cool things for people, like
iptel, fwd, etc.

If there is SIP paradise, our set up is most definitely nearer to hell,
regarding nat, because no one knows which kind of address the asterisk
client is going to come up with.

The more I fiddle with asterisk and read this list, the bigger my doubts
about the possibility of making asterisk (SIP) work for most of us (it
already works for some).

A friend suggested that maybe putting up one or two asterisk boxes to
work and using SER in strategically choosen nodes we could get away with it.

I'm having a look at SER and think that maybe it could work for us, but
wanted to check with some other people before diving into the unknown.

Answers like Give it a try,  Don't even think of it or Better back
to tam-tam and smoke signalling are wellcome.

Thanks for your time.

-- 
Pizco Dominguez
--

--
GPGKEY: gpg --keyserver pgp.rediris.es --recv-key 8DE37A4D
FINGERPRINT:85CB 4323 F322 5837 EDB5  2033 6FB2 C326 8DE3 7A4D
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--
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Re: [Asterisk-Users] zaphfc troubles

[Nicolas Olivier]

Stuart,

I switched the system to a pentium based host, with different memory.
The results are the same. I've also changed the ISDN card to be sure.

Nicolas

Stuart Hirst wrote:
 Nicolas,
 
 I replied earlier stating that I saw similar issues and now that you
 have applied the Florz patch the symptoms you are seeing are all but
 identical to the issues I saw and resolved by changing out the
 motherboard memory. The system was an ASUS main board with a Xeon
 processor.
 
 It is not the memory it could be something specific to the VIA motherboard.
 
 Stuart
 
 
 
 Nicolas Olivier wrote:
 
Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded
 after ztcfg with:

May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
 underrun: 0, 0
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
 overflow: 311, 311
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
 overflow: 436, 436
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
 underrun: 0, 0

And when I start asterisk, same stuff, kernel crashes.

Interrupts are ok.

sjaak imap wrote:
 

Dear Nicolas Olivier

Just try the florz patch at http://zaphfc.florz.dyndns.org/
and look at cat /proc/interupts if your not sharing irq's

Maybe this will help


Good luck

Sjaak

   

Hi,

I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with
 zaphfc.

The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e

When I'm doing the insmod on zaptel, zaphfc, zaprtc:

Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
 

0xc2d58000(0x2d58000) IRQ 12 HZ 100
   

zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e

I'm using zaprtc as the gateway is running on a VIA motherboard without
 

USB controller.
   

When I'm doing ztcfg -vv:

Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)

3 channels configured.

Here are my confs:

/etc/zaptel.conf:

loadzone=fr
defaultzone=fr

span=1,1,3,ccs,ami
bchan=1-2
dchan=3

/etc/asterisk/zapata.conf:

[channels]

language=fr
context=test
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
immediate=yes
channel = 1-2

/etc/asterisk/modules.conf:

[modules]
autoload=yes

noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so

noload = app_intercom.so

load = chan_modem.so
load = res_features.so
load = res_musiconhold.so
load = chan_zap.so

noload = chan_alsa.so
noload = chan_oss.so

[global]
chan_modem.so=yes
chan_zap.so=yes


The problem is that after ztcfg ran, I've got the following logs:

Registered tone zone 2 (France)
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 layer 1 state = F3
zaphfc: card 0 layer 1 state = F4
zaphfc: card 0 layer 1 state = F5
zaphfc: card 0 layer 1 state = F7
zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
 

wanted 8 got 7), probably a buffer overrun.
   

zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
 

wanted 8 got 7), probably a buffer overrun.
   

zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes

And when I start asterisk -c, same logs keep on, and I've finally a
 

kernel crash:
   

Unable to handle kernel paging request at virtual address fffc
printing eip:
c0113cc0
*pde = d063
*pte = 
Oops: 
CPU:0
EIP:0010:[c0113cc0]Not tainted
EFLAGS: 00010013
eax: c248015c   ebx:    ecx: 0001   edx: 0001
esi: c24803a0   edi: c248015c   ebp: c2c8fe2c   esp: c2c8fe14
ds: 0018   es: 0018   ss: 0018
Process sshd (pid: 146, stackpage=c2c8f000)
Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
 

c3819545
   

   0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4
 

0086
   

   c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008
 

c270c800
   

Call Trace:[c3819545] [c01cb6b1] [c381aae6] [c381aad7]
 

[c383cd78]
   

 [c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd]
 

[c0109f78]
   

 [c010c328]

Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - 

[Asterisk-Users] Polycom Instant Messaging

[Chris Coulthurst]

Can anyone explain the Polycom
Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to
it? Ive seen vague references to
MSN Messenger, and somehow thats mentally disturbing



Chris Coulthurst

[EMAIL PROTECTED]










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RE: [Asterisk-Users] VoiPSupply Dot Com

[mr. barker]

. Snip

It is sad to hear that you will not be purchasing from us. I do not
understand though, why we owe you an explanation for our toll free number
being down.

^^
You are right you don't owe any explanation at all for your numbers being
down.  It was your Toll Free and Your Toll number! Not just your Toll Free. 

For me personally once I find a company to deal with I usually stick with
them.  I have ordered in an excess of $3000.00 from your company.  When I
did call last week I tried on numerous ocassions and had the same result.
I even call the information and there was no listing for B2 Technologies nor
VoipSupply (this doesn't mean much though as you must subscribe to the
listing service).  What strikes me very odd is that I tried to call the
numbers from work and from home same results no longer in service.  I also
called through the my VOIP provider (which routes down to the US) and the
local teleco here.  I wanted to make sure that something was not up so I
asked my brother to try placing a call (he lives in a different province)
and he had the same no long in service.
This is not just one isolated incident involving one call!

I was going to post something last week but decided to not as I wanted to
see how your response was in the community.  As I predicated there would be
someone posting a something to the effect about how is VoipSupply to deal
with then followed by people saying that the service is reliable ... etc.

Now I may be just a bit over cautious when it comes to dealing with internet
based businesses because of being burnt before along with 1000's of others.

This is just my 2cents.



Lastly, we do charge for technical support. We are hear to help, but the
low margins on ATA's etc certainly does not leave us room to give away
free support. All of you that are ITSP's know exactly what I am talking
about.


If you order something, and you can't get it to work, you can pay for us
to make it work for you. If you order the wrong product, then that is your
mistake not ours.

There is an open invite to all to call or email me at any time to discuss
or business. Constructive criticism is always welcomed.

Thank you all for business and we look for more in the future!

Garrett Smith
VoIPSupply.com
[EMAIL PROTECTED]
716-250-3408 Direct


 mr. barker wrote:

 I tried calling their toll free number and toll number last week in
 the morning and afternoon and was handed a recording saying this
 number is no longer in service. The web site was up but there was no
 message on the site as to why the phone numbers were not working.

 I just called the number now and it is working.

 Being around the internet for a quite a long time this gives me an
 uneasy feeling. I have seen company's start to go under and pull the
 plug when they get into financial trouble(not being able to pay the
 bills) and run with the customers money. I have had this happen to me
 on 2 occasions. Just the woes of doing business on the net.

 Being in Canada it makes it very difficult to find companies that will
 ship COD from the US. If I was to order I would only order COD from
 now on from VoipSupply.

 I have ordered product from VoipSupply and received the product. I
 will not be ordering more product do to this outage of the phones with
 no explanation.

 Just my 2cents.

 Maybe the tollfree provider was responsible for the outage and maybe it
 only affected service from Canada.

 They accept credit cards and paypal. I believe you would have some
 recourse if they ran with your money.

 I quit shipping anything COD to anywhere a few years ago. If the
 customer refuses delivery the vendor loses money. When UPS instituted a
 policy of not handling cash payment for COD, I quit for good.

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[Asterisk-Users] listening at 5070

[Kamran Ahmad]

hello

sip.conf
bindport=5070

i am trying to register at ser 5060. but why i am
getting request at asterisk 5070.

thanks
Kamran





Yahoo! Mail
Stay connected, organized, and protected. Take the tour:
http://tour.mail.yahoo.com/mailtour.html

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[Asterisk-Users] HP Proliant ML110 with Adaptec 2610SA and Debian

[Alex]

Hi guys,
I am trying to install Debian sarge (latest netinstall) on ML110 server  
with two SATA hardware mirrored drives on Adaptec 2610SA controller for  
use with Asterisk with no luck.

Debian installer does not see the array. Any workarounds?
Please help.
Regards,
Alex.
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RE: [Asterisk-Users] Guest

[Nabeel Jafferali]

 For example, how does your dialplan look on the zap and sip servers in
 order
 to route the call from a zap on server 1 to a sip on server 2?

If you want any SIP server/client to be able to call you at
[EMAIL PROTECTED], for example, then in the context that is set in the
[general] part of sip.conf (usually default), add something like:

[default]
exten = anton,1,Goto(internal,200,1)

Similarly, if you want a specific server to be able to do this, add a peer
entry for that server that sets the context, and in that context put
something like the above.

Then, on that server, you would Dial(SIP/[EMAIL PROTECTED]).

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Anton Krall
 Sent: May 18, 2005 1:28 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Guest
 
 
 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of
 |Jason Walker
 |Sent: Martes, 17 de Mayo de 2005 11:41 p.m.
 |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 |Subject: RE: [Asterisk-Users] Guest
 |
 |I am a newbie to *, but if the far end of the call has no
 |route to your phone, how do you think this could be accomplished?
 |
 |I have agents log into one SIP server (no ZAP cards, just
 |SIP). Calls come through another * box with ZAP cards that are
 |routed to the SIP only server via the extensions.conf file.
 |
 |It seems to me that the far end would need something in their
 |dialplan to allow for calls to an extension to go to your SIP server.
 |
 |I apologize if I am giving a newbie response - I am also in
 |the process of learning.
 |
 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of
 |Anton Krall
 |Sent: Tuesday, May 17, 2005 9:08 PM
 |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 |Subject: [Asterisk-Users] Guest
 |
 |Guys.
 |
 |What do I need to configure in order to let my Asterisk
 |receive calls from sip phones, etc not registered with my
 |server on my extension?
 |
 |For example, let people use their asterisks or sip phones to
 |call [EMAIL PROTECTED]
 |
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 |
 |
 |--
 |No virus found in this outgoing message.
 |Checked by AVG Anti-Virus.
 |Version: 7.0.308 / Virus Database: 266.11.11 - Release Date: 5/16/2005
 |
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RE: [Asterisk-Users] SIP/nat situation

[Alex Vishnev]

Pizco,

SER is definitely better suited to deal with NAT issues then ASTERISK is. I
suggest looking at SER and NAT helpers like media proxy application (part of
SER). I also recommend looking at NAT devices at SER wiki page to make sure
that your router/nat device is compatible. In general, this is doable, but
will require a lot of playing around to get it right. There are a lot of
threads on both SER and ASTERISK wiki site to get both working nicely
together. 

Asterisk/SER Wiki Site www.voip-info.org

HTH

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pizco
Dominguez
Sent: Wednesday, May 18, 2005 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP/nat situation

Hi.

We are trying to set up asterisk to service a wireless community in our
town.

We have about 30/40 wireless working nodes each one with a 10.34.x.x/24
subnet for users. Each one of these addresses can potentially have a
192.168.x.x/x subnet.

On top, the wireless nodes, themselves, are linked in 172.16.x.x/x
subnets.

On top of the top, there is internet and cool things for people, like
iptel, fwd, etc.

If there is SIP paradise, our set up is most definitely nearer to hell,
regarding nat, because no one knows which kind of address the asterisk
client is going to come up with.

The more I fiddle with asterisk and read this list, the bigger my doubts
about the possibility of making asterisk (SIP) work for most of us (it
already works for some).

A friend suggested that maybe putting up one or two asterisk boxes to
work and using SER in strategically choosen nodes we could get away with it.

I'm having a look at SER and think that maybe it could work for us, but
wanted to check with some other people before diving into the unknown.

Answers like Give it a try,  Don't even think of it or Better back
to tam-tam and smoke signalling are wellcome.

Thanks for your time.

-- 
Pizco Dominguez
--

--
GPGKEY: gpg --keyserver pgp.rediris.es --recv-key 8DE37A4D
FINGERPRINT:85CB 4323 F322 5837 EDB5  2033 6FB2 C326 8DE3 7A4D
--

--
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Re: [Asterisk-Users] FWD to Asterisk stops after 3 seconds

[Michael Graves]

Sounds like reinvite troubles. Once the SIP endpoints are both in the
call the server (FWD) will get out of the way allowing the two SIP
clients to connect directly. There can be cases where you can connect
through the server but not directly, usually because of NAT traversal
failure at one end or the other. 

Are you connecting to FWD through SIP or IAX?

Michael

On Wed, 18 May 2005 18:49:49 +0800, Ronald Wiplinger wrote:

I asked my friend to setup FWD and call me to my *

However, it did not matter which codec we used, after three seconds the 
connection was cut.

Why? and how to make it stabled?


bye

Ronald

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--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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Re: [Asterisk-Users] Cisco contract for 7940/7960 firmware access

[Eric Wieling aka ManxPower]

Mark Johnson wrote:
Mark Brown wrote:
Hi Everyone!
Is there any hope for us newbie plebs who want to also get hold of the 
updated Cisco firmware?

I need to get a 7910G updated to work on SIP..
Any help on obtaining the updated firmware quickly and painlessly 
would be great J

Cheers
M
7910 does not have a SIP image and looks like it never will. I have 
about 40 of these stupid things that I can't get to work 100% with 
skinny or sccp. If you ever figure out how, be sure to let me know!
There are some references to SIP for the 7910 on the Cisco web site. 
They are wrong.  As Mark said, there is no SIP firmware for the 7910.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] Audio flutter on OH323 output?

[Tony Mountifield]

Hi, I'm using OH323, mostly with success, to interface Asterisk to
a provider's switch (World Telecom INX). I have noticed a particular
effect, and I wonder whether anyone else has seen the same?

The effect is audio flutter (almost like the flutter one gets on
MF or HF radio sometimes) which only happens intermittently.
Audio coming into Asterisk is unaffected, as proved by using the
Monitor app as follows:

Phone1-PSTN-Switch-(via H.323)-Asterisk(Monitor+DISA)-Switch-PSTN-Phone2.

Intermittently, each party hears the other party's audio flutter for a few
seconds. Reviewing the recordings made by Monitor, no flutter is present,
so the incoming audio is fine.

Note that this is a direct call. I've also noticed it on MeetMe, where
it seems again that the flutter is on the audio leaving Asterisk.
Different participants may hear the flutter at different times.

The system is a dual-Xeon 3GHz running Fedora Core 3 with the STABLE
branch of Asterisk from CVS, together with oh323 0.6.5, openh323 1.13.5.3
and pwlib 1.6.6.3.

Any suggestions would be appreciated!

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] sip show registry empty ?!?!!?

[Eric Wieling aka ManxPower]

Michele O-Zone Pinassi wrote:
Hi all,
i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) 
and this is what my sip show users return:

moloch*CLI sip show users
Username Secret   Accountcode Def.Context ACL  NAT
204  moirafrom-internal   No   No
203  michele  from-internal   No   No
202  duccio   from-internal   No   No
201  fabrizio from-internal   No   No
moloch*CLI

it's ok. So i use kphone to connect top my asterisk server. KPhone say that 
i'm on-line so i'll check sip show registry and it's empty:

moloch*CLI sip show registry
HostUsername   Refresh State
moloch*CLI
sip show registry shows remote systems Asterisk is registered to.
Try sip show peers
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] Integrating Asterisk into our Legacy PBX --Newb

[Geoff Manning]

I have been successful in setting up asterisk and making workstation to
workstation SIP calls. But I am lost when it comes to anything past that.

We are trying to integrate this asterisk server into with our Executone
(432?) PBX to allow us to make outbound SIP calls between our disparate
locations.

We have a T1 card in our PBX, and the Digium TE110P card in the Asterisk. We
have the T1 card connected to a CSU and the CSU going into the TE110P. And
the Asterisk server connected to the WAN.

We get dial tone from the T1 card but that's as far as I get! Where do I
begin?

Thanks in advance!!


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[Asterisk-Users] Accessing Voice Mail

[Christopher Kenna]

Ihave Ext 101 configured as the default for incoming calls. Ext 101 also holds all of the incoming voicemails. How do I access the voicemail for ext 101 remotely? I am lookingto be able to call in from the outside and retrieve all of my messages. When I press *97 during the voicemail outgoing message, it only prompts me to change to a different extension though the directory list.

Chris

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[Asterisk-Users] Asterisk and rfc2833 help

[James Bushey]

Hi All,
Im having some trouble getting Asterisk to send DTMF via rfc2833.  The 
scenario is this:

For purposes of testing software, I have two applications communicating 
with each other via DTMF.  In between the two applications sits an 
Asterisk.  The applications require that DTMF be sent via rfc2833, 
otherwise they will not understand the DTMF.

The first application (app-a) makes an outbound call to the second 
application (app-b) via SIP through the Asterisk instance.  The Asterisk 
then fully sets up the SIP call and bridges the two applications.  The 
problem is that when DTMF being sent via rfc2833 reaches Asterisk, 
asterisk then puts this DTMF in-band and sends it via the RTP stream.  
Sending DTMF through the other three legs of the call are just fine.  
That is, from app-a to asterisk, and in both directions between asterisk 
and app-b, dtmf is sent via rfc2833, its just this one section of the 
call that goes in-band.

Heres a simple diagram for clarification:
appa  asterisk   
  app-b
-rfc2833--  
---rfc2833---
-in-band--  
-rfc2833-

Also note that I can, and sometimes do, substitute a GrandStream 
BudgetTone-100 SIP phone in place of app-a, and the problem persists.

When removing the Asterisk instance from the middle of the call, app-a 
is fully able to communicate with app-b via rfc2833.
Any help would be greatly appreciated.  Let me know if you have any 
questions or need some clarifications.

Thanks,
~James
--
James Bushey
Software Engineer
Soleo Communications
ph: 585-641-4300 x0050
fax: 585-641-0502
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Re: [Asterisk-Users] VoipSupply.com

[Michael Graves]

On Wed, 18 May 2005 00:01:53 -0400, Paul wrote:

Manjit Riat wrote:

 I am going to buy some IP phones from them but I sent them an email 
 couple of weeks ago and got no reply. Has anyone ordered anything from 
 them? Any other places that I can buy from? Sorry if it’s a wrong post.

Not getting a reply to email is definitely the exception with them. They 
reply to emails. When you call they sometimes ask for your email address 
and send you additional information. I look at product information on 
manufacturer websites. If I am still interested, I then look for 
possible vendors. Voipsupply seems to carry almost all of the voip 
products that interest me. I only hear positive feedback from people who 
have purchased goods from them. They offer extended warranty/replacement 
coverage on many products. Need I say more? Yes - they all seem like 
very nice people.

I heartily agree. I just recently purchased a Wifi SIP phone from them.
For a small order they too the time to walk me through the details
about the device. When I called the sales rep back with a question I
got voice mail. He called me back inside an hour.

Highly recommended.

Michael
--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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Re: [Asterisk-Users] how to get remote extensions to work correctly with a zap channel?

[Eric Wieling aka ManxPower]

Jon Gabrielson wrote:
I am trying to get remote extensions to work correctly with
agents.  I have ackcall=yes and have agents logged in to 
extension 101 using agentcallbacklogin with extension 101 defined as:

exten = 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer))
This setup works great on local and/or voip channels, but on zap 
channels, the zap channel answers immediately as soon as it goes off hook 
and the announcement gets played long before the agent gets a chance 
to answer their phone.

Is there a way to either delay the announcement until the agent picks up
or to keep repeating the message until the agent presses a button?
Or is there possibly a better way of doing this?
Incorrect!  Only ANALOG zap channels are considered answered as soon 
as the digits are finished dialing.  There is a 'c option to help 
with this, but it's not well documented.  Dial(Zap/3c/5551212).

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] DHCP, PoE, FXS, FXO and ONE power adapter ONLY???

[Michael Graves]

On Wed, 18 May 2005 11:45:51 +0100, Iqbal wrote:

doesnt invetel do one

Iqbal

Ronald Wiplinger wrote:

 This afternoon we were discussing, and found that we would like one 
 box, which should have ALL of these:

 1. WAN port
 2. Ethernet port 1 with Power over Ethernet
 3. Ethernet port 2 with or without PoE
 4. FXS port
 5. FXO port
 6. DHCP, web configureable.
 7. Optional wireless accesspoint
 8. One and ONLY one power adapter for this box

 Does such a box exist?

Almost sounds like my new Astlinux server with a TDM 400 and a second
network cardif that second NIC could inject POE. Astlinux includes
router with traffic shaping as well, NTP server, and Asterisk as well.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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Re: [Asterisk-Users] HP Proliant ML110 with Adaptec 2610SA and Debian

[Matteo Brancaleoni]

mmh I think you asked to the wrong ML,
this is Asterisk, not Debian installer ML.

Cya.

On Wed, 2005-05-18 at 23:00 +1000, Alex wrote:
 Hi guys,
 
 I am trying to install Debian sarge (latest netinstall) on ML110 server  
 with two SATA hardware mirrored drives on Adaptec 2610SA controller for  
 use with Asterisk with no luck.
 
 Debian installer does not see the array. Any workarounds?
 
 Please help.
 
 Regards,
 Alex.
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-- 
Matteo Brancaleoni
System Administrator
Tel  +39.02.70633354
Sip  [EMAIL PROTECTED]
Iax2 [EMAIL PROTECTED]

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RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)

[asterisk]

MSN will only work on 1 ISDN2 line and cannot be spread across 2 ISDN2
lines. From your description I assume you have 2 calls up and the 3rd call
fails. This is because you can only have 2 concurrent calls using MSN on
ISDN2. You will find you have a different number range for the second ISDN2
If you want to use both ISDN lines for incoming calls with the same number
range then you will need to have the lines converted to 1 + 1 Auxiliary
working and have the numbers delivered as DDI.

Neil

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Norvall
Sent: 18 May 2005 13:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)

Hi

I can see what seems to be both devices in use, so I guess it must be
down to the capi.conf (below), does this look correct ???

[interfaces]

msn=292880
incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886,
292887, 292888, 292889
outgoingmsn=292880
controller=1
softdtmf=1
;accountcode=
context=demo
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2

msn=292xxx
incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886,
292887, 292888, 292889
outgoingmsn=292880
controller=2
softdtmf=1
;accountcode=
context=demo
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: 18 May 2005 12:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)

On Wed, 18 May 2005, Lee Norvall wrote:
 Hi
  
 I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server.  We
can
 use all 4 lines for out going calls fine, but on incoming we can only
use 2.
 On calling in using the main msn, the 3rd line gives a an engaged
signal.
 
 I have unplugged 1 of the cards, and the other card takes the 2 calls.
I
 then swapped this around, and this also works fine.  But when using
both
 cards, we can only use 2 line in.

There are two possibilities:

1) your Telco doesn't send the 3rd call to your other line.
   You can verify that by using
divactrl mlog -c 1 -o  (diva_idi module must be leaded)
   and see if an incoming call is shown.
   (use -c 2 for the second card)

2) your configuration of chan_capi is not correct and the 3rd call is 
   ignored/rejected.

If you don't use DIVA Server cards with CAPI, forget this mail.

Armin

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[Asterisk-Users] Wrong password on authentication for NOTIFY

[c waddy]

Hi,

I am trying to get to the bottom of a warning i am recieving through
the console.

May 18 13:26:29 WARNING[8281]: chan_sip.c:6837 handle_response:
Forbidden - wrong password on authentication for NOTIFY

Calls are still working. I cannot work out what is causing it.

Asterisk - Ingate - Asterisk.

I have googled and cannot find anything on the above.

Thanks.
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Re: [Asterisk-Users] VoipSupply.com

[Ed Greenberg]

--On Tuesday, May 17, 2005 5:24 PM -0700 Manjit Riat [EMAIL PROTECTED] 
wrote:


I am going to buy some IP phones from them but I sent them an email
couple of weeks ago and got no reply. Has anyone ordered anything from
them? Any other places that I can buy from? Sorry if it's a wrong post.
I have never had a problem with a VoipSupply order - both on the web and on 
the phone.

I recommend writing directly to Garrett at [EMAIL PROTECTED] with your 
inquiry.

/edg
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RE: [Asterisk-Users] Call forwarding...

[Kanuri, Seshu (Company IT)]

Try changing SetCIDNum SetCallerID and use to SetCIDName as under:

Ex:
---
exten = s, 1, SetCallerID(${CALLERIDNUM})
exten = s, 2, SetCIDName(${CALLERIDNAME})
exten = s, 3, Dial(${ARG2}/${ARG1},${RINGSECS})
exten = s, 4, Voicemail(u${ARG1})
exten = s, 5, Hangup
exten = s, 101, Voicemail(b${ARG1})
exten = s, 102, Hangup
 
Seshu Kanuri


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Benson
Sent: Wednesday, May 18, 2005 6:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call forwarding...

Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...

Hi,

I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.

I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based - no real phone lines).

I tried this (from voip-info.org wiki)...

exten = 1234,1,dial(sip/1234,20)
exten = 1234,2,playback(pls-wait-connect-call)
exten = 1234,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 1234,4,SetCIDNum(0${CALLERIDNUM}) exten =
1234,5,dial(${TRUNK}c/9871234321,20,r)
exten = 1234,6,SetCIDNum(${NewCaller})
exten = 1234,7,voicemail2([EMAIL PROTECTED]) exten =
1234,101,voicemail2([EMAIL PROTECTED])
exten = 1234,102,hangup

Mine looks like this...

exten = 08700688nnn,1,Dial(SIP/operator,1,t)
exten = 08700688nnn,2,playback(pls-wait-connect-call)
exten = 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 08700688nnn,4,SetCIDNum(0${CALLERIDNUM})
exten = 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r)
exten = 08700688nnn,6,SetCIDNum(${NewCaller})
exten = 08700688nnn,7,Voicemail(u100)
exten = 08700688nnn,8,Hangup()
exten = 08700688nnn,101,Voicemail(b100) exten =
08700688nnn,102,Hangup()

(where nnn is a real number)
The sip channel is set to time out quickly for testing.
And I don't appear to have the pls-wait-connect-call audio file - but
that isn't an issue for the time being...
The IAX2/0870n is the extention/device that calls go out on via
voiptalk... (my call provider)...
If I include the c/ in the TRUNK line I get...

   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
c/07961106nnn|20|r) in new stack
May 18 10:37:40 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for 'c'
May 18 10:37:40 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)

Asterisk shows this from the moment the sip channel is considered not to
have answered (1 sec)...

   -- Nobody picked up in 1000 ms
   -- Executing Playback(IAX2/[EMAIL PROTECTED]:4569-1,
pls-wait-connect-call) in new stack
May 18 10:20:26 WARNING[24416]: file.c:486 ast_openstream_full: File
pls-wait-connect-call does not exist in any format May 18 10:20:26
WARNING[24416]: file.c:790 ast_streamfile: Unable to open
pls-wait-connect-call (format ilbc): No such file or directory May 18
10:20:26 WARNING[24416]: app_playback.c:83 playback_exec: 
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569-1 for
pls-wait-connect-call
   -- Executing SetVar(IAX2/[EMAIL PROTECTED]:4569-1,
NewCaller=01202843nnn) in new stack
   -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1,
001202843nnn) in new stack
   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)
   -- Executing SetCIDNum(IAX2/[EMAIL PROTECTED]:4569-1,
01202843nnn) in new stack
   -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]:4569-1,
u100) in new stack
   -- Playing '/var/spool/asterisk/voicemail/default/100/unavail' 
(language 'en')

Again - I'm not worried about the audio file warning - I can fix that
later... I guess this is the important bit...

   -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-1,
/07961106nnn|20|r) in new stack
May 18 10:20:26 WARNING[24416]: channel.c:1957 ast_request: No channel
type registered for ''
May 18 10:20:26 NOTICE[24416]: app_dial.c:927 dial_exec_full: Unable to
create channel of type '' (cause 66)  == Everyone is busy/congested at
this time (1:0/0/1)

The call then drops into voicemail...

I've tried various permuations but still no call is made to the mobile
number. Any ideas?

Cheers,

Mark

I should mention that I have tried using the call forward function of
the sip phones, but a) this means configuring the phones and some are
remote and behind firewalls and b) It doesn't work... 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does 
not waive confidentiality or privilege, and use is prohibited. 
 
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Re: [Asterisk-Users] zaphfc troubles

[Peer Oliver Schmidt]

Nicolas Olivier wrote:
Quoting from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation
As I haven't got a Digium card, I need a timer which can be provided by 
ztdummy, zaprtc or zaprai.
But anyway the results are the same with or without zaprtc loaded.
Irregardless of your problem, the ZAPHFC cards do provide the timer 
needed for MOH, IAX trunking etc.
--
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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[Asterisk-Users] Mysql cmd with Asterisk Problems

[pbx]

Hello all:
I am trying to use the mysql command to retrieve information from a mysql
database.
my example here was formed from using the wiki reference to using the
mysql command.

The problem is with the fetch command.
Here is the macro code:
Mysql(QueryString=SELECT\ ivr-password\ from\ users\ where\ ivr-id=${userid})
Mysql(Query r ${connid} ${QueryString})
Mysql(Fetch fetchid ${r} dbuserpass)
Mysql(Clear ${resultid})
Mysql(Disconnect ${connid})

However, it never gets past the fetch line. and ${r} is not showing
anything either from che CLI window.
I usesd the mysqlasteri web page to make the command escape character
happy, etc. I have tried putting \' around each item, etc. However The
same problem comes back with the fetch line. I have tried to use
mysql(fetch fetchid ${r} ivr-password) thinking the variable that is
coming out of the DB has to be named the same, etc. but it doesn't matter.
${r} is blank in the fetch command when I know there is a valid record.
The ${connid} has a value in it as it's being passed.

I have only been able to find the mysql cmd example on the wiki, and no
other. I program mysql with php all the time, but i dont understand the
errors that it is returning..
Thank you.

Output is below:

-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-1, )
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-1, )
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-1, Authenticate user
now: userid: 1234 - pass: )
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-1, Passing to
macro-mmisd-login)
-- Executing Macro(IAX2/[EMAIL PROTECTED]:4569-1,
mmisd-login|1234|)
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-1, Entered
Macro-mmisd-login)
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569-1, Passed in userid:
1234 - userpass: )
-- Executing MYSQL(IAX2/[EMAIL PROTECTED]:4569-1, Connect connid
localhost mmisd-ivr-user mmisd-ivr-pass ivr-db)
-- Executing SetVar(IAX2/[EMAIL PROTECTED]:4569-1,
QueryString=SELECT\ ivr-password\ from\ users\ where\ ivr-id=1234)
-- Executing MYSQL(IAX2/[EMAIL PROTECTED]:4569-1, Query r 18
SELECT\ ivr-password\ from\ users\ where\ ivr-id=1234)
-- Executing MYSQL(IAX2/[EMAIL PROTECTED]:4569-1, Fetch fetchid 
dbuserpass)
May 18 07:01:05 WARNING[7114]: app_addon_sql_mysql.c:113 find_identifier:
Identifier 0, identifier_type 2 not found in identifier list
May 18 07:01:05 WARNING[7114]: app_addon_sql_mysql.c:328 aMYSQL_fetch:
aMYSQL_fetch: Invalid result identifier 0 passed


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[Asterisk-Users] RTFriendsCache=yes help Voicemail MWI help

[pbx]

A while back I converted back to static conf files from a database setup.

However I decided to tackle it again.

The problem that I was experiencing, was, there was no stutter tone on my
sipura 2000 or 3000 when there was a voicemail left at either extension
when I was using mysql setup for peers and voicemail.

I have 2 contexts... home, office in my voicemail configuration
I now use VoicemailMain([EMAIL PROTECTED]) context being office, or home.. and
that all works.

It's just the stutter tone that does not work.

Someone suggested, and i also found in the wiki to place
rtfriendscache=yes in the sip.conf file, and I have tried that as well.
Still no avail.

In the mysql record for the registration i have in the mailbox field i.e
[EMAIL PROTECTED] (the sipura extension is 2000) and the context is home. but
still no stutter tone.

If I use the static voicemail.conf file and sip.conf file the stutter tone
works.

Thanks in advance

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Re: [Asterisk-Users] Asterisk and Ericsson PBX

[Niksa Baldun]

I am unsure of what you want to achieve. Do you want to interconnect BP
and Asterisk, or replace BP with Asterisk? What is the purpose of
proprietary software you mention? Please give more details.

Niksa



[EMAIL PROTECTED] wrote:

Hi, I´m trying to migrate my propietary software to an asterisk server 
connected to a Ericsson BP 128i PBX.

I´ve been looking at the asterisk web, user forums, published docs about how 
to use the PBX as the hardware device but I haven´t found anything.

I think this is possible. The old server is currently connected to the 
Ericcson via serial port.

Please, help.

Thanks a lot.



This message is for the designated recipient only and may contain privileged, 
proprietary, or otherwise private information.  If you have received it in 
error, please notify the sender immediately and delete the original.  Any 
other use of the email by you is prohibited.
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[Asterisk-Users] Softphone Requirements

[Bill Ford]

Has anyone seen a Softphone with the following features:

1) Utilizes Touch Screen
2) Has API for interfacing CID info with existing application on same PC.


Thanks
Bill Ford
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RE: [Asterisk-Users] Static on TDM Zaptel FXO

[Gregory Wiktor - ADCom Corp.]

Hello Rod, I'll try it, thanks.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Wednesday, May 18, 2005 1:01 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Static on TDM Zaptel FXO

Make sure you have disabled framebuffer, apic and acpi.


--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
==
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Re: [Asterisk-Users] Asterisk - Spandsp: fax header

[Jean-Yves Avenard]

Hi PeterOn 18/05/2005, at 10:05 PM, Steve Underwood wrote:It is only there because the sending machine put it there in the image. Spandsp is not different from how any FAX machine I have ever used behaves. As well as sending the 20 digit number as text, the sending machine puts in the header. This is not what I'm referring to... I know what is being put by the remote fax !On my Brother's fax machine (MFC-8820D) today, I've received 3 faxes: all of them at the top showed the caller Fax identity.I received 2 faxes on Asterisk with spandsp, one from the same sender as earlier on the brother: there's nothing at the top.I wouldn't ask if it was obvious the data was inside the image, give me some credits for God's sake !Typically, when somebody is sending a fax on the Brother unit, once the connection has been established the identity of the fax caller is then displayed on the Brother's LCD (and this has nothing to do with PSTN CallerID), what is displayed on the LCD will be printed at the top of each pages. This is this behavior I'm trying to reproduce with Asterisk/Spandsp.JY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200  ___
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[Asterisk-Users] Best Compression Available

[Michael Stearne]

Hi,

What would you say that the best compression format is for voice
recordings on Asterisk?  The tradeoff being the file's size.  I like
GSM because of the small files size but the quality isn't great.  What
are people finding as a good setting for GSM?

Thanks,
Michael
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RE: [Asterisk-Users] Static on TDM Zaptel FXO

[Gregory Wiktor - ADCom Corp.]

Hello Bryce,
Gain settings do seem to have an effect.  I am going from a Cisco
7960AsteriskZap TDM CardPOTS 

Thanks,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryce
Chidester
Sent: Wednesday, May 18, 2005 1:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO

That type of echo is usually caused by incorrectly (or not at all) tuned
gain settings in zapata.conf. I don't know what kind of phones you're
using, but for Asterisk to even be able to detect DTMF tones on our
Sayson / Aastra 390s and 480s, our FXS channels are set to -5.0 on both
rx and txgain. If you're using externally-powered phones (as in not your
ordinary joe-schmoe analog phone), I have found that they're usually
pretty hot (loud) and Asterisk can't understand what is said.
Good luck!

Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:  
[EMAIL PROTECTED]/305


On May 17, 2005, at 19:00, Gregory Wiktor - ADCom Corp. wrote:


 On a recompile of the kernel I now get a 99.98 average.
 Static is gone, although quality so far seems not quite there yet.

 I am also experiencing an odd local echo.  I can hear a slight echo 
 locally, but the other end sounds fine, and the other end does not get

 echo.

 Even with the pots disconnected, you can hear it.  The static would be

 on all calls.  Hooking up a normal phone was ok.  The sipsip phones 
 are perfect too, it was only happening on the zap channel...

 Greg

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy 
 Parr
 Sent: Monday, May 16, 2005 8:05 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO

 On 5/16/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote:


 Hello All,
 I recently put in a zaptel 1fxo/1fxs card.  I am experiencing heavy 
 static.

 Even with the pots line disconnected, if I do a dial I still get


 static.


 This way I know it's not the line, but rather something on the card.

 I tried alternate pci slots.

 This card has a power connector, does anyone know what the power 
 requirements are?  The unit is in a small case with a 2.4ghz p-4 and 
 512mb ram, on an intel board with 533fsb.  All other functions are


 fine.



 I am using the latest CVS on Debian 2.6test

 Anyone experience this?



 Have you tried a different phone? Does the static appear immediately 
 when you pick up the phone? Or on the second or third time?
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[Asterisk-Users] Re: SigSeg in channel.c / chan_mISDN problem ?

[Andreas Czerniak]

Hmm, i can re-produce this problem in a way:
- external call to voip
- voip terminate this call
After this, asterisk produce an sigseg like:
I SEND:DISCONNECT   port:1  pid:0   mode:TE addr:51400101
-- l3id:20011 cause:16 dad:72 oad:xyxyxyxyxyxyxyxyxy channel:1 port:1
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault (core dumped)
Can be this a problem with the chan_mISDN driver ?
Thx,
Andreas.
--On Mittwoch, 18. Mai 2005 15:10 +0200 Andreas Czerniak [EMAIL PROTECTED] 
wrote:

Dear !
After an update from 1.0.3 - 1.0.7 Asterisk, I have an Segmentation
fault at regular intervals in the channel.c file. Every SigSeg produce an
core dump file. After loading this in gdb, asterisk interrupt every time
in the same line:
# 0  0x0805dac6 in ast_queue_frame (chan=0x81bcb48, fin=0x41203750) at
# channel.c:384
384 cur = chan-pvt-readq;
Our configuration: Asterisk 1.0.7, on a linux 2.6.11.9 with mISDN and
CAPI devices.
Have anyone an hint for more debugging output or a solution for this
problem ?
Thx in advanced,
Andreas.
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RE: [Asterisk-Users] Integrating Asterisk into our Legacy PBX -- Newb (correction)

[Geoff Manning]

Correction:

The hardware is a Wildcard T100P (not a TE110P)

Thanks!

 -Original Message-
 From: Geoff Manning [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, May 18, 2005 9:07 AM
 To: Asterisk Users (E-mail)
 Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX
 --Newb
 
 
 I have been successful in setting up asterisk and making 
 workstation to
 workstation SIP calls. But I am lost when it comes to 
 anything past that.
 
 We are trying to integrate this asterisk server into with our 
 Executone
 (432?) PBX to allow us to make outbound SIP calls between our 
 disparate
 locations.
 
 We have a T1 card in our PBX, and the Digium TE110P card in 
 the Asterisk. We
 have the T1 card connected to a CSU and the CSU going into 
 the TE110P. And
 the Asterisk server connected to the WAN.
 
 We get dial tone from the T1 card but that's as far as I get! 
 Where do I
 begin?
 
 Thanks in advance!!
 
 
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[Asterisk-Users] ASTERISK-SIPP

[Biagio Meirone]

Someone say of configure sipp
with asterisk and asterisk with sipp

I have a lot of problem for
sdp






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