Re: [asterisk-users] TDM400 FXO stopped working

[Remco Barendse]

On Mon, 26 Sep 2011, Michael L. Young wrote:



I think the clue is actually right there in the error message.

You say that port 1 is an FXO module?  Then your signaling is set wrong.  The 
signaling should be fxsks.

For port 4, it should be fxoks.

Remember, that in the configuration files, the signaling option used is 
opposite of what the module is.

Regards,
Michael Young
(elguero)



I think somehow the module is not recognized. Switching the signalling 
types doesn't help, i get the same error.  Running dahdi_genconf generates 
this :


# Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
# channel 1, WCTDM/4/0, no module.
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
fxsks=4
echocanceller=mg2,4


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Re: [asterisk-users] TDM400 FXO stopped working

[Remco Barendse]

Hi Vladimir,

I tried the steps as you wrote them but i just get the same error :
Channel map:

Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)

2 channels to configure.

Changing signalling on channel 1 from Unused to FXO Kewlstart
DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
Selected signaling not supported
Possible causes:
FXO signaling is being used on a FXO interface (use a FXS 
signaling variant)

RBS signaling is being used on a E1 CCS span
Signaling is being assigned to channel 16 of an E1 CAS span


The config i used has been working for years, i wonder what changed.

Thanks!!

Best regards, Remco

On Sat, 24 Sep 2011, Vladimir Mikhelson wrote:


Remco,

In my experience something could have gone wrong with the parameters you pass 
to WCTDM driver.

You may want to check your /etc/modprobe.d/dahdi.conf

Try loading the driver manually as follows:

1. Stop Asterisk
2. modprobe -r wctdm
3. modprobe wctdm
4. dahdi_cfg -vvv

Good luck,
Vladimir



On 9/23/2011 4:27 AM, Remco Barendse wrote:

Hi list

I have 2 servers with a TDM400 card, port 1 populated by an FXO (red)
module), port 4 populated with an FXS module. I am using dahdi linux
and tools 2.5.0.1. The servers are running CentOS 4 and the other box
CentOS 6.

Both modules have been working fine but recently stopped working, when
i start dahdi with just FXS enabled everything is fine.

This is the error i get :
Loading DAHDI hardware modules:
  wctdm:   [  OK  ]

Running dahdi_cfg:  DAHDI_CHANCONFIG failed on channel 1: Invalid
argument (22)
Selected signaling not supported
Possible causes:
FXO signaling is being used on a FXO interface (use a FXS
signaling variant)
RBS signaling is being used on a E1 CCS span
Signaling is being assigned to channel 16 of an E1 CAS span
   [FAILED]


This is in my system.conf :
fxoks=1
echocanceller=mg2,1
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
fxsks=4
echocanceller=mg2,4

# Global data

loadzone= nl
defaultzone = nl


When i run dahdi_genconf it doesn't detect the module either :
# Autogenerated by /usr/sbin/dahdi_genconf on Fri Sep 23 11:24:16 2011
# Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
# channel 1, WCTDM/4/0, no module.
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
fxsks=4
echocanceller=mg2,4


I already replaced both FXO modules with new ones but without result.

Ideas anyone?

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[asterisk-users] TDM400 FXO stopped working

[Remco Barendse]

Hi list

I have 2 servers with a TDM400 card, port 1 populated by an FXO (red) 
module), port 4 populated with an FXS module. I am using dahdi 
linux and tools 2.5.0.1. The servers are running CentOS 4 and the other 
box CentOS 6.


Both modules have been working fine but recently stopped working, when i 
start dahdi with just FXS enabled everything is fine.


This is the error i get :
Loading DAHDI hardware modules:
  wctdm:   [  OK  ]

Running dahdi_cfg:  DAHDI_CHANCONFIG failed on channel 1: Invalid argument 
(22)

Selected signaling not supported
Possible causes:
FXO signaling is being used on a FXO interface (use a FXS 
signaling variant)

RBS signaling is being used on a E1 CCS span
Signaling is being assigned to channel 16 of an E1 CAS span
   [FAILED]


This is in my system.conf :
fxoks=1
echocanceller=mg2,1
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
fxsks=4
echocanceller=mg2,4

# Global data

loadzone= nl
defaultzone = nl


When i run dahdi_genconf it doesn't detect the module either :
# Autogenerated by /usr/sbin/dahdi_genconf on Fri Sep 23 11:24:16 2011
# Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
# channel 1, WCTDM/4/0, no module.
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
fxsks=4
echocanceller=mg2,4


I already replaced both FXO modules with new ones but without result.

Ideas anyone?

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[asterisk-users] Asterisk RPM repo?

[Remco Barendse]

I am reinstalling a server and wanted to give the asterisk rpm's a try.

It seems however that the repo on asterisk.org doesn't know anything more 
recent than RHEL / CentOS 5.


Is there a more recent repo?

Will asterisk work with selinux?

Thanks!

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[asterisk-users] Shorewall rate limiting rules?

[Remco Barendse]

Reading of all the brute force attacks on the list i was wondering if 
anyone has implemented some connection rate limiting rules in Shorewall to 
stop the brute force attacks?

I'm a bit puzzled about which rule(s) to use on which ports, if 
anyone could help me with some example rules to start with it would be 
great :)

Thanks!

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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

[Remco Barendse]

On Sun, 11 Apr 2010, Mark Smith wrote:

>
> Same this end from 184.73.17.150.
>
> Use this little piece of iptables magic to block the whole of Amazon's EC2 ip-
> range.
>
> iptables -F
> iptables -A INPUT -m iprange --src-range 216.182.224.0-216.182.239.255 -j DROP
> iptables -A INPUT -m iprange --src-range 72.44.32.0-72.44.63.255 -j DROP
> iptables -A INPUT -m iprange --src-range 67.202.0.0-67.202.63.255 -j DROP
> iptables -A INPUT -m iprange --src-range 75.101.128.0-75.101.255.255 -j DROP
> iptables -A INPUT -m iprange --src-range 174.129.0.0-174.129.255.255 -j DROP
> iptables -A INPUT -m iprange --src-range 204.236.192.0-204.236.255.255 -j DROP
> iptables -A INPUT -m iprange --src-range 184.73.0.0-184.73.255.255 -j DROP
> iptables -A INPUT -m iprange --src-range 216.236.128.0-216.236.191.255 -j DROP
> iptables -A INPUT -m iprange --src-range 184.72.0.0-184.72.63.255 -j DROP
> iptables -A INPUT -m iprange --src-range 79.125.0.0-79.125.127.255 -j DROP
> service iptables save
>
> This sorts it out in the short-term until Amazon realise their service is
> being utilised by arseholes.
>

Would this work if using Shorewall? What would a sane ruleset for 
Shorewall look like that implements some sort of rate limiting features?



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[asterisk-users] Intel Atom based Asterisk server?

[Remco Barendse]

I currently have some Asterisk home servers on general pc hardware as well 
as a mission critical server asterisk pbx running on a Dell 2850

To reduce noise and power consumption i would like to migrate them all to 
an Intel Atom based solution, showstoppers so far were single NIC and 
single PCI slot motherboards. I found that Supermicro makes a Dual NIC 
board with one PCI slot and 2 PCI-Express slots (X7SLA-L)

Has anyone tried running Asterisk + CentOS 5 on this (or any other) 
Atom board? Is the Atom platform able to handle the load of all the 
interrupts a TE110P or TDM400P card will generate ?

I am aware about other solutions but i do use the servers for some other 
tasks therefore don't want to move to a dedicated pbx box based on Soekris 
or the likes.

Thanks for any input!

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Re: [asterisk-users] Dahdi/callerid issue

[Remco Barendse]

On Mon, 18 Jan 2010, ev...@disruptor.nl wrote:

> Hey Ira,
>
> It seems after a several testing, that the wait(1) seems to solve the issue.
>
> Only now weirdly enough the phone keeps ringing if the caller hangs up
> before i picked up the phone (pstn call)
>
> Regards,
>
> Evert

Hi Evert

I have been suffering the same problem ever since the switch from Zaptel 
-> DAHDI, already exchanging DSL line filters and everything.

Where did you add the wait and can you post a snippet of your config line?

I spent hours on this trying to figure it out but couldn't (big thanks to 
you for doing the work for me) ;)

Remco

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[asterisk-users] DNS reload on trunks for outgoing calls

[Remco Barendse]

Is there any fix or workaround for the DNS problem (old standing bug that 
when the box starts and domain names do not resolve quickly enough from 
DNS then asterisk stops using the outgoing trunks.

I read on the list before that it is considered a huge and unacceptable 
load for asterisk servers to try and resolve the domain names again 
after some time but it is rather annoying. I don't know about 
resources of other people but on my boxes i have some cpu cycles that 
could be used for that :)

I now do nightly restarts of asterisk but it still means that at least for 
one day calls are flowing through expensive PSTN.

If anybody knows of a workaround, would be most welcome

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[asterisk-users] Example to handle incoming calls without callerid at home?

[Remco Barendse]

I am using asterisk 1.6 at home and would like to send incoming calls 
without caller id immediately to voicemail (i don't want to use the 
privacy manager where people have to enter a number).

The config examples i found are all for the pretty obsolete 1.0 and 1.2 
versions of asterisk.

Would anyone be willing to share a config example?

Thanks!

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[asterisk-users] sip show channels shows non-existent channels on 1.6.0.19 and 1.4.27.1 ?

[Remco Barendse]

I never do sip show channels but i tried it this morning to see if 
everything is working after upgrading 2 boxes to 1.6.0.19 and 1.4.27.1

Is it correct that Asterisk doesn't clean up sip channels anymore after 
using them ?

On one box i can see sip lines for every phone that was attempted to call, 
on the other box i see sip channels for 2 sip peers i register too. This 
is the output on the console:

1.4.27.1 :
pbx*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Format Hold   Last Message
1.1.1.1  mysipname   576687fc4a8  00573/0  0x0 (nothing)  No
10.0.0.0 221 774a1331284  00102/0  0x0 (nothing)  No 
Init: NOTIFY
10.0.0.1 224 0359f0d23a4  00102/0  0x0 (nothing)  No 
Init: NOTIFY

etc.

1.6.0.19 :
Peer User/ANRCall ID  Format   Hold 
Last Message
8.0.0.0 (None)  d34db33f-125982  0x0 (nothing)No 
Rx: OPTIONS
8.0.0.0 (None)  d34db33f-125982  0x0 (nothing)No 
Rx: OPTIONS
2 active SIP dialogs


Ip addressed modified (nuked) for the above examples

Is this normal?

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Re: [asterisk-users] Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11 Now Available

[Remco Barendse]

Thanks for this new release :)

Just out of curiosity, why did the download page for asterisk.org change?

In the old days it was quite clearly visibible what the latest asterisk 
version was including all the related packages for that version 
(libpri, addons, dahdi etc. etc.)

Why not do something similar now?  asterisk-addons have completely 
disappered from the page, one now needs to dive into the news releases 
or the ftp to find the latest and greatest.

Why not split it up like it was before, like :

Asterisk-1.4.27.1   asterisk-1.6.0.19
dahdi-linux-2.2.0.2 dahdi-linux-2.2.0.2
dahdi-tools-2.2.0   dahdi-tools-2.2.0
libpri-1.4.10.2 libpri-1.4.10.2
asterisk-addons-1.4.9   asterisk-addons-1.4.9
skypeforasterisk-1.4_1.0.6  skypeforasterisk-1.6.0_1.0.6
whatever-1.4.0.4whatever-1.6.0.3

Just a thought :)

Cheers!
Remco

On Mon, 30 Nov 2009, Asterisk Development Team wrote:

> The Asterisk Development Team has announced the release of Asterisk 1.2.37,
> 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate
> download at http://downloads.asterisk.org/pub/telephony/asterisk/
>
> These releases have been created in response to a SIP remote crash
> vulnerability.
>
> Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain 
> an
> SDP regression fix as described in issue #16268.
>
> Asterisk 1.6.0.19, and 1.6.1.11 contain an additional SDP regression fix as
> described by issue #16238.
>
> Information about the SDP issues can be found at
> https://issues.asterisk.org/view.php?id=16268 and
> https://issues.asterisk.org/view.php?id=16238
>
> For more information about the details of this vulnerability, please read the
> security advisory AST-2009-010, which was released at the same time as this
> announcement.
>
> The security advisory is available at
> http://downloads.asterisk.org/pub/security/AST-2009-010.pdf
>
> For a full list of changes in the current releases, please see the ChangeLogs:
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.2.37
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.27.1
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.19
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.11
>
> Thank you for your continued support of Asterisk!
>
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Re: [asterisk-users] SNOM 870

[Remco Barendse]

On Mon, 2 Nov 2009, SIP wrote:

> That's odd. We've had Snom 190s, 320s, and 360s running day in day out
> for years with not a single issue. Maybe we got all the good ones from
> your batch. If that's the case, I thank you for 'taking one for the
> team' as it were. ;)

Perhaps i am REALLY unlucky (or just the batch i bought is really from the 
beginning when there were still issues to be solved)

A summary so far of phones I RMA'ed already (i bought 36 Snom 360's in 
2005) :

5 x Receiver hook broken
3 x Display broken
1 x 360 completely died altogether
2 x power supplies (i switched to PoE to cover that one)

About the receiver hook, it is not the actual switch that breaks, it is 
the plastic "arms" inside the phone that are supposed to keep the hook 
switch in place that get broken. With some care and glue it is possible to 
glue it back together but still.

I have another batch now ready to send for for repair, 3 power 
supplies, 2 phones with a broken hook switch and 2 more where the display 
is not working anymore.

Rather disappointing numbers for a business phone i would say. I also 
bought 7 units of the 190 which have worked flawlessly

I did not include the receive cord as a malfunction of the phone, i 
consider that normal wear & tear

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Re: [asterisk-users] SNOM 870

[Remco Barendse]

On Fri, 30 Oct 2009, hbk wrote:

> Hi,
> 
> I have played with the 820 for some weeks, mostly love it excellent speech 
> quality. Even got the "mini" browser running
> showing my favorite webcam, this could be put to real use too:)
> 
> Issues so far:
> Some embarrassing crashes while speaking, was able to speak but all freezed. 
> Still a little fresh firmware I guess.
> Error 404 after showing webcam picture, but it works!
> Have to use *1 to start recording, record soft button does not seem to work 
> with *.
> 
> Still I recommend it, best IP phone I have tried!
> Not sure 870 is worth the extra money, not tested that yet.

How is the build quality of the 870?

The mortality rate on power supplies, diplays and the number or broken 
receiver hook swicthes on the lot of Snom 360's i bought 3 years ago is 
outright embarrassing.



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[asterisk-users] Missing digits from CallerID on TDM400P?

[Remco Barendse]

I have a TDM400P hooked up to an analog line from KPN in The 
Netherlands. CallerID is working but sometimes some digits are missing 
from the number, i.e. if the number that calls me is:
0204569236

I will sometimes get this in the display:
020456236

Which digit is missing seems to be fairly random. Where / how could i 
start debugging this ?

This is my etc/dahdi/system.conf :
fxoks=1
echocanceller=mg2,1
fxsks=4
echocanceller=mg2,4

loadzone= nl
defaultzone = nl


This is my /etc/asterisk/chan_dahdi.conf :
echocancel=yes
echocancelwhenbridged=yes
echotraining=400

callerid=202
signalling=fxo_ks
group=1
context=intern-all
channel=>1

signalling=fxs_ks
cidsignalling=dtmf
cidstart=polarity
usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
group=2
context=inbound-analog
channel=>4


Thanks!
Remco

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Re: [asterisk-users] Skype for Asterisk callfile question

[Remco Barendse]

On Wed, 2 Sep 2009, Matt Riddell wrote:

> On 2/09/09 7:45 PM, Remco Barendse wrote:
>> So i create a callfile that looks like this:
>> ---
>> Channel: SIP/228
>> MaxRetries: 0
>> Dial(Skype/asterisk...@somebodyonskype)
>> Priority: 1
>> Callerid: Somebodyonskype
>
> You're combining technologies there :)

Not hindered by any knowledge i was trying to get things working :)

Thanks, it seems to make sense to Asterisk now :)

The first time i configured SFA to allow incoming calls only, reloading 
the module does not allow outbound calls still (direction is not mentioed 
as a directive for which asterisk needs to be restarted but it doesn't 
really matter).

Thanks again for your help.

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[asterisk-users] Skype for Asterisk callfile question

[Remco Barendse]

Hi list,

To make outgoing calls by skype i would like to have our crm app create 
callfiles like we do for normal calls.

If i read the instructions it says this :
---quote---
The syntax for making an outgoing call using Skype for Asterisk is as 
follows:
Dial(Skype/[@])
---unquote---


So i create a callfile that looks like this:
---
Channel: SIP/228
MaxRetries: 0
Dial(Skype/asterisk...@somebodyonskype)
Priority: 1
Callerid: Somebodyonskype 
---
SIP/228 is my desk phone, i purposely did not include a context (is it 
necessary for Skype calls? i guess not because i also did not create a 
'dial plan' for it, should be just dump and go i guess)

But i guess something must be wrong on my dial string :
[Sep  2 09:39:10] NOTICE[8834]: pbx_spool.c:255 apply_outgoing: Syntax 
error at line 3 of /var/spool/asterisk/outgoing/REMCO.CALL
[Sep  2 09:39:10] WARNING[8834]: pbx_spool.c:260 apply_outgoing: At least 
one of app or extension must be specified, along with tech and dest in 
file /var/spool/asterisk/outgoing/REMCO.CALL
[Sep  2 09:39:10] WARNING[8834]: pbx_spool.c:427 scan_service: Invalid 
file contents in /var/spool/asterisk/outgoing/REMCO.CALL, deleting
[Sep  2 09:39:10] WARNING[8834]: pbx_spool.c:482 scan_thread: Failed to 
scan service '/var/spool/asterisk/outgoing/REMCO.CALL'

Where am i going wrong?

Thanks!!

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Re: [asterisk-users] Skype for Asterisk???

[Remco Barendse]

On Wed, 19 Aug 2009, Terry Wilson wrote:

> I haven't seen (or heard of) it happening.  Please post a bug report
> on http://betareports.digium.com/mantis/ with a backtrace from one of
> the core dumps along with the relevant information about your setup,
> dialplan, chan_skype.conf, etc.  If there is a crash, I need to fix
> it. :-)

I tried to install 
/asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.1-x86_32.tar.gz
instead of 1.4_1.0.0 but this version causes asterisk to segfault 
immediately after starting, with the previous version i had 
occasional segfaults, now asterisk never starts.

I wanted to setup Skype for Asterisk on a production PBX where segfaults, 
downtime and frequent reboots are a nuisance ofcourse.

I'll try to go down the debug path but the options to take down the 
production box for debugging are somewhat limited :)

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Re: [asterisk-users] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory

[Remco Barendse]

On Fri, 21 Aug 2009, Olivier wrote:

> So basically it's harmless, unless you actually have such a card.
> 
> 
> Yes, but as you mentioned, most don't have a transcoder card.
> My opinion is such message shouldn't be send at all for those environments 
> where there is no transcoder card, (as it will
> remain, IMHO, normal behaviour to care about ERROR messages).

I agree, or at worst reduce it to a message that just notifies that no 
transcoder card was found. When i see error i always think i did something 
wrong :)

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[asterisk-users] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory

[Remco Barendse]

I have a CentOS release 4.7 box running asterisk-1.4.26.1 with 
dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0

I regularly get these messages, is this something i should be worried 
about?

[Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so => (ITU 
G.726-32kbps G726 Transcoder)

[Aug 21 01:05:07] ERROR[4343] codec_dahdi.c: Failed to open 
/dev/dahdi/transcode: No such file or directory

[Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_dahdi.so => (Generic DAHDI 
Transcoder Codec Translator)

How to fix this ? Google was not my friend this time :)

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Re: [asterisk-users] Skype for Asterisk???

[Remco Barendse]

On Wed, 19 Aug 2009, Terry Wilson wrote:

>>> Have you posted a bug describing the issues you are having at 
>>> http://betareports.digium.com/mantis/
>>> yet? I would love to have the opportunity to actually fix any bugs
>>> that people find.  :-)
>>
>> I installed the 1.0 release of Skype for Asterisk and last night on my
>> production box running Asterisk 1.26.1 i got segfaults and 32 core
>> dumps,
>> all happened in a time frame between 01:04 - 01:08 at night (so 4
>> minutes).
>>
>> Anyone else seeing this?
>
> I haven't seen (or heard of) it happening.  Please post a bug report
> on http://betareports.digium.com/mantis/ with a backtrace from one of
> the core dumps along with the relevant information about your setup,
> dialplan, chan_skype.conf, etc.  If there is a crash, I need to fix
> it. :-)

I never used Skype myself but i installed it to try and i noticed that i 
got added by lots of strange skype users (spam bots?), i guess some of 
those were trying some funny stuff on my skype for asterisk account. I 
want to use Skype for Asterisk only for incoming calls at this time.

Is there a how to "filing bug reports for dummies" ?

I am running Asterisk 1.4.26.1 with the latest FreePBX and Skype for 
Asterisk is the only add-on. I did *not* enable the G.729 codec in the 
config neither did i try to install the G.729 codec license that comes 
with Skype for Asterisk (nowhere it said that the G.729 is required for 
correct operation of Skype and i don't care about the bandwidth at this 
time).

The core dumps i can just delete, they are useless?

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Re: [asterisk-users] Skype for Asterisk???

[Remco Barendse]

Oops sorry, the Asterisk version should read 1.4.26.1


On Wed, 19 Aug 2009, Julian Lyndon-Smith wrote:


Nope - but you are also running on an unsupported version of asterisk,
so I am not surprised. From the readme:

===[ Installation Overview ]===

It is required that the proper version of Asterisk is installed prior to
installing Skype For Asterisk. Skype For Asterisk is currently supported on:

  Asterisk 1.4 versions >= 1.4.25
  Asterisk 1.6.0 versions >= 1.6.0.6
  Asterisk 1.6.1 versions >= 1.6.1.5

Previous versions of Asterisk WILL NOT work properly with Skype For Asterisk.
It is also important to make sure that the major version of Skype For Asterisk
downloaded matches the version of Asterisk installed on the system. Trying to
compile Skype For Asterisk 1.4 versions on Asterisk 1.6.0 while fail, etc.
There is no version of Skype For Asterisk for Asterisk trunk.

Julian

2009/8/19 Remco Barendse :

On Tue, 18 Aug 2009, Terry Wilson wrote:


That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.


Have you posted a bug describing the issues you are having at 
http://betareports.digium.com/mantis/
 yet? I would love to have the opportunity to actually fix any bugs
that people find.  :-)


I installed the 1.0 release of Skype for Asterisk and last night on my
production box running Asterisk 1.26.1 i got segfaults and 32 core dumps,
all happened in a time frame between 01:04 - 01:08 at night (so 4
minutes).

Anyone else seeing this?

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Re: [asterisk-users] Skype for Asterisk???

[Remco Barendse]

On Tue, 18 Aug 2009, Terry Wilson wrote:

>> That does sound a bit pricey, although it it's as stable as the latest
>> beta, I wont be buying it at all.
>
> Have you posted a bug describing the issues you are having at 
> http://betareports.digium.com/mantis/
>  yet? I would love to have the opportunity to actually fix any bugs
> that people find.  :-)

I installed the 1.0 release of Skype for Asterisk and last night on my 
production box running Asterisk 1.26.1 i got segfaults and 32 core dumps, 
all happened in a time frame between 01:04 - 01:08 at night (so 4 
minutes).

Anyone else seeing this?

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Re: [asterisk-users] Skype for Asterisk???

[Remco Barendse]

On Mon, 17 Aug 2009, Pascal Bruno wrote:

> Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
> 
> $66 per channels, pretty pricey
> 
> http://store.digium.com/productview.php?product_code=1SFA0001

Yes, pretty pricey indeed especially considering that you can buy Skype 
ATA adapters for the same amount (or less).

But then again, who needs Skype for business purposes anyways, i don't 
think there is a huge market for it.

I will add one channel to our PBX and will see if anybody will call us 
using skype.

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Re: [asterisk-users] Asterisk 1.6.1 and dahdichanname = no

[Remco Barendse]

On Thu, 18 Jun 2009, Kevin P. Fleming wrote:

> Remco Barendse wrote:
>> I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk
>> 1.6.1.
>>
>> As FreePBX only supports ZAP naming i set dahdichanname = no in my
>> asterisk.conf.
>>
>> However, after installation the console was still merrily chattering about
>> incoming calls on DAHDI channels and nothing happened because all the
>> ZAP stuff was ignored.
>>
>> Are all Asterisk versions (1.6.0, 1.6.1 as well as the soon to be released
>> 1.6.2) able to deal with dahdichanname = no ?
>>
>> If yes, where could i be going wrong?
>
> No. Only Asterisk 1.4.x releases support 'dahdichanname'; 1.6.x releases
> require DAHDI, and will only support DAHDI channel names. If you saw any
> documentation to the contrary please point it out so we can get it fixed.

Thanks for your reply! Strictly speaking, i didn't. I found some posts on 
a forum that FreePBX works with Asterisk 1.6.x so i just assumed that it 
would work.

My bad, thanks for clearing it up!

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[asterisk-users] Asterisk 1.6.1 and dahdichanname = no

[Remco Barendse]

I am using FreePBX with Asterisk 1.4 and i wanted to upgrade to Asterisk 
1.6.1.

As FreePBX only supports ZAP naming i set dahdichanname = no in my 
asterisk.conf.

However, after installation the console was still merrily chattering about 
incoming calls on DAHDI channels and nothing happened because all the 
ZAP stuff was ignored.

Are all Asterisk versions (1.6.0, 1.6.1 as well as the soon to be released 
1.6.2) able to deal with dahdichanname = no ?

If yes, where could i be going wrong?

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Re: [asterisk-users] IP phone recommendation

[Remco Barendse]

On Wed, 3 Jun 2009, Rob Hillis wrote:

> Christian Stredicke wrote:
>> Check out the snom 300 or the snom 820...
>>
>
>
> Good lord... talk about two extremes... :)  The Snom 300 is pretty good,
> but the 320 is much better and costs around a *third* of what the Snom
> 820 does.
>
> Stick with the older model snoms.  So far I've seen nothing about the
> 820 to justify the significant extra expense.

Build quality perhaps? I hope the newer models last longer than 2 
years.


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Re: [asterisk-users] Asterisk w/ Nokia "e" Series Handsets

[Remco Barendse]

On Tue, 12 May 2009, Andrew Joakimsen wrote:

> Overall, given the limitations of WiFi, it works rather well. I've
> never had to reboot my E71 or play with the settings after it was
> setup. Something I can't say about other WiFi (only) phones I have
> used. And VoIP on Windows mobile phones is crap.

I installed a program called WeFi on my phone. To the phone it appears as 
one single access point, while WeFi handles connections to all access 
points automatically. It solves the problem of creating one SIP profile 
for each access point.


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Re: [asterisk-users] Rusting Snoms?

[Remco Barendse]

On Sat, 9 May 2009, Tim Panton wrote:

> This is a bit off topic, because I 'think' it isn't an Asterisk problem.
> However I'm not sure and anyhow I'm hoping someone may recognize the symptom.
>
> We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years 
> old)
> were packed up for the move, then unpacked a couple of weeks later.
>
> On unpacking them and connecting them to the new network, several of them
> didn't work well. The symptom is that outgoing RTP audio is garbled - like 
> the
> packets are pulsed. Inbound is fine. This isn't true for all of the phones,
> just some of them. (The all run the same SNOM firmware)

I didn't experience these problems, even when not using the phones for 
more than a year. The Snom 190s are pretty well built though, unlike 
the 360's that are just a piece of crap. Did you check the network 
settings, maybe there is still some NAT stuff present in the phones? I 
would do a factory reset on the phone and try again.

I bought 30 Snom 360's in 2005, since then 12 of them have been showing 
problems, an absolutely unacceptable mortality rate for a business phone.

I had 4 power supplies gone dead, of 4 phones the display went dead 
and i had about 7 phones where the receiver hook switch broke off inside 
the case (if you open the phone you see that the receiver hook switched is 
locked in between to tiny pieces of plastic). The hook switch even broke 
in my phone and i am very careful with it, i don't handle it rough...
(i'm not counting the power supplies among broken phones)

So far Snom repaired 2 batches under warranty, one after the 
warranty expired, the 3rd batch of 4 phones if waiting to be sent to 
them, will wait to see their reply, the fail rate is unacceptable and way 
beyond normal.

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Re: [asterisk-users] Faxing success rate on PRI

[Remco Barendse]

On Sun, 8 Mar 2009, benoit wrote:

>
> Here is my current setup:
>
>E1 => [Asterisk with TE220p] => IAX Trunk (routed network) =>
> [Asterisk with TDM800p] => Fax/Copy Machine

The TE220P and the TDM800P are in different Asterisk boxes? Any particular 
reason for that?  I now have an E1 coming in to the asterisk box and 
IAXmodem and HylaFAX to receive faxes which works flawlessly.

Outbound i still have an old faxserver connected to an old analogue line 
though.

My faxing needs require that i have a hardcopy printout of every fax that 
is sent like a reduced size of the fax sent on one A4 together with all 
status info (Fax number it was sent to, status, duration, ID of the 
receiver, date, time etc.). So far i couldn't find any solution that does 
that.


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Re: [asterisk-users] Compile problems

[Remco Barendse]

On Sun, 8 Mar 2009, Tzafrir Cohen wrote:

> On Sun, Mar 08, 2009 at 12:26:00AM +0100, Remco Barendse wrote:
>
>> So far so good but then when i do :
>> cd /usr/src/asterisk-1.6.0.6
>> make clean ; ./configure ; make all ; make install
>>
>> i get this :
>>
>> In file included from app_dahdiras.c:50:
>> /usr/include/dahdi/user.h:736: error: syntax error before "__s32"
>> /usr/include/dahdi/user.h:743: error: field `params' has incomplete type
>> /usr/include/dahdi/user.h:939: error: syntax error before "__s32"
>> /usr/include/dahdi/user.h:940: error: syntax error before ':' token
>> make[1]: *** [app_dahdiras.o] Error 1
>> make: *** [apps] Error 2
>
> Fixed in
> http://svn.digium.com/view/asterisk?view=revision&revision=178306

/n00b mode on

And how do i get that fix? :) Do i need to build asterisk from SVN, if yes 
how do i get the right version?

svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk

or something else?

/n00b mode off

By the way, the error seems to be fixed in asterisk-1.4.22.1 too 
:)


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Re: [asterisk-users] Compile problems

[Remco Barendse]

On Sun, 8 Mar 2009, Sebastian wrote:

> The fax error seems to be problem of spandsp version.
> What version are you using???

I use the latest IAXMODEM 1.2.0, the changelog of it says "update spandsp 
to 20080725 snapshot"

However, i never asked Asterisk to compile with fax support, can i disable 
fax support somewhere?  make menuconfig for asterisk didn't get me much 
further

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[asterisk-users] Compile problems

[Remco Barendse]

Hi all

I don't know what went wrong but i no longer seem to be able to compile 
asterisk. I first do :

cd /usr/src/dahdi-linux-2.1.0.4
make clean ; make all ; make install

cd /usr/src/dahdi-tools-2.1.0.2
./configure ; make clean ; make all ; make install ; make config

So far so good but then when i do :
cd /usr/src/asterisk-1.6.0.6
make clean ; ./configure ; make all ; make install

i get this :

In file included from app_dahdiras.c:50:
/usr/include/dahdi/user.h:736: error: syntax error before "__s32"
/usr/include/dahdi/user.h:743: error: field `params' has incomplete type
/usr/include/dahdi/user.h:939: error: syntax error before "__s32"
/usr/include/dahdi/user.h:940: error: syntax error before ':' token
make[1]: *** [app_dahdiras.o] Error 1
make: *** [apps] Error 2

When i try asterisk 1.6.0.5 i get :
[CC] app_fax.c -> app_fax.o
app_fax.c: In function `transmit_audio':
app_fax.c:344: error: structure has no member named `t30_state'
app_fax.c:347: error: structure has no member named `t30_state'
app_fax.c:348: error: structure has no member named `t30_state'
app_fax.c:349: error: structure has no member named `t30_state'
app_fax.c:353: error: structure has no member named `t30_state'
app_fax.c:418: error: structure has no member named `t30_state'
app_fax.c:420: error: structure has no member named `t30_state'
app_fax.c:459: error: structure has no member named `t30_state'
app_fax.c:462: error: structure has no member named `t30_state'
app_fax.c: In function `transmit_t38':
app_fax.c:498: error: structure has no member named `t30_state'
app_fax.c:499: error: structure has no member named `t38'
app_fax.c:502: error: structure has no member named `t30_state'
app_fax.c:503: error: structure has no member named `t30_state'
app_fax.c:504: error: structure has no member named `t30_state'
app_fax.c:506: error: structure has no member named `t30_state'
app_fax.c:532: error: structure has no member named `t38'
app_fax.c:535: error: structure has no member named `t30_state'
app_fax.c:537: error: structure has no member named `t30_state'
app_fax.c:567: error: structure has no member named `t30_state'
make[1]: *** [app_fax.o] Error 1
make: *** [apps] Error 2


And when i try 1.6.1-rc1 i get :
[CC] app_fax.c -> app_fax.o
app_fax.c: In function `transmit_audio':
app_fax.c:331: error: structure has no member named `t30_state'
app_fax.c: In function `transmit_t38':
app_fax.c:511: error: structure has no member named `t30_state'
app_fax.c:512: error: structure has no member named `t38'
make[1]: *** [app_fax.o] Error 1
make: *** [apps] Error 2


This is my system info :
CentOS 4.7  with :
Linux version 2.6.9-78.0.13.ELsmp (mockbu...@builder16.centos.org) (gcc 
version 3.4.6 20060404 (Red Hat 3.4.6-10)) #1 SMP Wed Jan 14 16:12:46 EST 
2009

configure: Package configured for:
configure: OS type  : linux-gnu
configure: Host CPU : i686
configure: build-cpu:vendor:os: i686 : pc : linux-gnu :
configure: host-cpu:vendor:os: i686 : pc : linux-gnu :


Asterisk always used to compile without probs?? Any tips / helps / 
pointers most welcome!

Remco

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Re: [asterisk-users] SIP *8 Pickup Problem

[Remco Barendse]

On Fri, 6 Mar 2009, Klaus Darilion wrote:
>
> Updating to 1.4 branch solved the issue. Thanks.

Pity that they still didn't release a new version that works properly.

1.6.0.6 is broken too, SIP doesn't work on 2 difference boxes i tried it 
on.

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[asterisk-users] Asterisk 1.6.0.6 sip doesn't work?

[Remco Barendse]

I tried upgrading to 1.6.0.6 but when i compile and install that, it seems 
that support for SIP is missing completely?

Reverting back to 1.6.0.5 gets SIP going again...


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Re: [asterisk-users] SMS /w Asterisk

[Remco Barendse]

On Tue, 10 Feb 2009, Steve Totaro wrote:

>> In that case, you would not need Asterisk at all.  If you can create
>> call files can you hit a URL from your CRM as well?

Not really, the app cannot open a browser, but it can create a file on a 
samba share quite easily.

Therefore going through asterisk seemed to be the way. Alternatively i 
could try and recreate something similar like call files for asterisk, 
have the crm app create a text file with the complete url in it and feed 
the url to a browser like links.  However with my skills at writing 
scripts being zero i thought that going through asterisk is the most 
obvious way for me.

> If you wanted to go through Asterisk, I would think a call file that
> drop into a context with a FastAGI to your CRM that returns the text
> would be my approach, or similarly, through the manager interface.

OK, thanks, i will throw all that in Google-o-matic and see what comes up 
:)


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Re: [asterisk-users] SMS /w Asterisk

[Remco Barendse]

On Tue, 10 Feb 2009, Steve Totaro wrote:

> Kannel is probably the best way to go in the States, unless you want
> to sign up with an aggregator.
>
> I use Kannel and a bank of Sony Ericsson phones.  To send SMS, you
> just have to hit a URL on the Kannel server with a properly formatted
> URL.  I just use System() to call Lynx with the correct variables (the
> message you want to send).

Just out of curiosity, how did you transfer the text you want to SMS to 
Asterisk? Can that be done through a call file? I want to SMS from a CRM 
app, the CRM app can create call files but for security reasons i do not 
want the sms server accessible from the network to each PC that can run 
the CRM app.

On voip-info there are some affordable GSM adapters that also provide an 
SMS server accessible by URL, i would like to do something similar like 
you did, would you be willing to post your configs?

Thanks a 1,000,000 :)

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Re: [asterisk-users] Broken Pipe error while using UpdateConfig command

[Remco Barendse]

1.4.23.1 is quite badly broken and there are no significant new 
features

Better to revert back to 1.4.22.1

On Tue, 3 Feb 2009, Jose P. Espinal wrote:

> Hello List,
>
> I have been working on a little PHP software that uses AMI's
> UpdateConfig command in order to modify some of it's config files.
>
>
> I was working with 'Asterisk 1.4.22.1' and everything was working.
> After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type:
>
>
> ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error:
> Broken pipe
>
>
> I'm completely sure that I did not modify anything on the PHP script,
> in fact, I test it on the older version of Asterisk mentioned above
> and it still works like a charm.
>
> Can someone point me out about a possible place to start looking for
> this error?
>
>
> NOTE:
> An interesting thing to note is that I sent all the commands that my
> script executes on the AMI to a file (/tmp/debug.txt )
>
> Then copying all the file content into AMI interface directly ( telnet
> [server_ip] 5038) and it executed all the commands without any problems.
>
>
>
> Thanks for your help,
>
>
> --
> Jose P. Espinal
> http://eslackware.com
>
>
>
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Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released

[Remco Barendse]

On Thu, 29 Jan 2009, Thomas Stein wrote:

> On Thursday 29 January 2009 09:23:41 Remco Barendse wrote:
>> 1.4.23.1 doesn't seem to work for me.
>>
>> I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest
>> zaptel as well. Incoming calls stopped working. Whenever an extension was
>> trying to pickup the phone by doing a group pickup with *8 the extension
>> just got dead audio and the next phone in the group stared ringing.
>
> Yeah. Thats http://bugs.digium.com:80/view.php?id=14206
>
> I'm also concerned about that one:
> http://bugs.digium.com:80/view.php?id=13488
>
> cheers
> t.

Thanks for your reply, indeed that is the problem. Strange that this 
"stable" release is still prominently on the asterisk.org website as the 
latest and greatest.

The latest bug you mentioned is only valid for mISDN installations i 
think?

Thanks again!

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Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released

[Remco Barendse]

1.4.23.1 doesn't seem to work for me.

I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest 
zaptel as well. Incoming calls stopped working. Whenever an extension was 
trying to pickup the phone by doing a group pickup with *8 the extension 
just got dead audio and the next phone in the group stared ringing.

I was running 1.4.23.1 with the latest FreePBX.

1.4.22.2 is working fine.

(Yes i know zaptel was replaced by DAHDI but upgrading is a PITA)


On Fri, 23 Jan 2009, Asterisk Development Team wrote:

> The Asterisk.org development team has announced the release of Asterisk
> 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for
> immediate download from http://downloads.digium.com/.
>
> This update for Asterisk includes a security fix for chan_iax2. Please see the
> associated security adivisory for more details:
> http://downloads.digium.com/pub/security/AST-2009-001.html
>
> These updates are a fix to a previous security release (released as versions
> 1.2.31, 1.4.22.1, and 1.6.0.3).
>
> The new versions are being released after additional testing revealed some
> issues with the way that scanning for users was blocked. Those issues have
> been corrected in this release.
>
> This security issue affects the 1.2, 1.4, and 1.6 series of Asterisk.
>
> Also note, that Asterisk 1.6.0.4-rc1 was released yesterday prior to the
> security update. That release has been removed as there will be no 1.6.0.4
> release, but rather will be reincarnated as 1.6.0.6-rc1. The reason for
> the dead release is to avoid 5 digit release numbers.
>
> ChangeLogs for the various releases are available at:
>
> http://downloads.digium.com/pub/asterisk/ChangeLog-1.2.31.1
> http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.22.2
> http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.23.1
> http://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.5
>
> Thank you for your continued support of Asterisk!
>
>
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Re: [asterisk-users] DAHDI trouble (again) Unable to open master device '/dev/zap/ctl'

[Remco Barendse]

On Sun, 18 Jan 2009, Tzafrir Cohen wrote:

>> Unloading DAHDI hardware modules: doneLoading DAHDI hardware modules:
>>wctdm:  Notice: Configuration file is /etc/zaptel.conf
>> line 0: Unable to open master device '/dev/zap/ctl'
>
>  grep ztcfg /etc/modprobe.conf /etc/modprobe.d/*
>
> Remove those lines. They are pointless.

Removing that seems to work, funny i have not been bitten by that before.

Thanks for your help!

Pity there is no uninstall script that chirurgically removes all old 
zaptel stuff from the system or at least a document with DAHDI that 
describes to remove what and where.



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[asterisk-users] DAHDI trouble (again) Unable to open master device '/dev/zap/ctl'

[Remco Barendse]

Because of a kernel upgrade i needed to recompile DAHDI. Dahdi 2.0.0 with 
2.0.1 was working ok, after the reboot and compile it doesn't start 
anymore.

Firstly it tells me it is using /etc/zaptel.conf which is deprecated, if i 
remove that file it complains that this file is missing.

Secondly i get an error about a missing dev file which refers to zap 
(which seems strange to me because i use DAHDI).

Where did i go wrong?  The box is Centos 4 with kernel 2.6.9-78.0.13.ELsmp



Unloading DAHDI hardware modules: doneLoading DAHDI hardware modules:
   wctdm:  Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for wctdm
[FAILED]

Running dahdi_cfg: [  OK  ]


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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Remco Barendse]

On Thu, 9 Oct 2008, Sean Bright wrote:

> On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse <[EMAIL PROTECTED]> wrote:
>> The information (or lack of it) on upgrading from zaptel to that
>> @&*^QW%&^%!!!  dahdi is very frustrating.
>>
>> I cannot find anything on how to uninstall zaptel, i found an earlier post
>> to this list which suggested make uninstall and make remove in the zaptel
>> directory which just generates errors and does nothing (on zaptel 12.1).
>
> What types of errors do you encounter running 'make uninstall'?
> You'll need to make sure both asterisk and zaptel are shutdown before
> running make install:
>
> # service asterisk stop
> # service zaptel stop

[EMAIL PROTECTED] dahdi-tools-2.0.0]# cd /usr/src/zaptel-1.4.12.1/
[EMAIL PROTECTED] zaptel-1.4.12.1]# make uninstall
make: *** No rule to make target `uninstall'.  Stop.
[EMAIL PROTECTED] zaptel-1.4.12.1]# make remove
make: *** No rule to make target `remove'.  Stop.
[EMAIL PROTECTED] zaptel-1.4.12.1]#

Looking through the makefile there is only a target for make 
uninstall-modules which ofcourse only removes part of zaptel, not the init 
scripts and all the other stuff

> Unfortunately there was a bug in the initial 2.0.0 release.  This has
> since been resolved in Subversion (see more details here
> http://bugs.digium.com/view.php?id=13615).
>
> If you'd like, you can grab the latest from Subversion of both the
> DAHDI Linux an DAHDI Tools packages, using the following commands:
>
> $ svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux
> $ svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools
>
>> Also the config files and everything are much more complicated
>> for dahdi than they were for zaptel
>
> As far as I am aware, the format of the configuration files
> (/etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf) are
> basically the same as their predecessors, /etc/zaptel.conf and
> /etc/asterisk/zapata.conf.  Feel free to post here with any questions
> and we'll try to help out.

OK, will do :) Thanks!


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Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Remco Barendse]

On Thu, 9 Oct 2008, Steve Totaro wrote:

> I don't have answers just a question.
>
> DAHDI is alpha or beta code, what motivates you to upgrade so badly that you
> are frustrating yourself so much?

Perhaps the fact that zaptel is not listed anymore on the Digium website? 
:)

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[asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

[Remco Barendse]

The information (or lack of it) on upgrading from zaptel to that 
@&*^QW%&^%!!!  dahdi is very frustrating.

I cannot find anything on how to uninstall zaptel, i found an earlier post 
to this list which suggested make uninstall and make remove in the zaptel 
directory which just generates errors and does nothing (on zaptel 12.1).

Then i install dahdi-linux and dahdi-tools and i want to start configuring 
it, so i am trying dahdi_genconf like the docs suggested which generates 
this really helpful error message :
/usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No 
such file or directory

Also the config files and everything are much more complicated 
for dahdi than they were for zaptel

There was some nice documentation and examples on how to get started with 
configuring certain devices with zaptel on the digium page, for my TDM11B 
they only mention zaptel.

Did anyone even try this?

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[asterisk-users] Zaptel -> DAHDI for dummies?

[Remco Barendse]

Is there an install script or step-by-step instruction somewhere on 
whaty is needed to migrate from zaptel to dahdi?

I read the document that digium published which nicely states some of
the differences between zaptel and dahdi but i was looking more for 
something like step-by-step instructions.

I'm sure i will not be the only one who had this question but the wiki and 
google didn't help me much (or maybe everybody is still sticking with 
zaptel for the time being).

Thanks!

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Re: [asterisk-users] Restrict SIP registration to one ip address only?

[Remco Barendse]

On Wed, 17 Sep 2008, Jared Smith wrote:

> On Wed, 2008-09-17 at 19:58 +0200, Remco Barendse wrote:
>> Why doesn't Asterisk allow both username&pass as well as setting an ip
>> adress on a sip.extension?
>
> It does.  To enforce ACLs on a SIP user or peer or friend, simply use
> "permit" and "deny" statements to allow and disallow various IP
> addresses or subnets.  Standard practice seems to be to deny everything
> first, then specifically allow other IP addresses.
>
> [user]
> type=friend
> secret=mypassword
> host=dynamic
> deny=0.0.0.0/0
> permit=10.1.2.3
> permit=192.168.123.0/24
> permit=192.168.222.0/255.255.255.0

Cool, this is exactly what i was looking for, i couldn't find a reference 
to it anywhere else.

Suprising that this feature isn't used much, i would suspect that many 
asterisk installations (including mine) have very simple (short) extension 
numbers which makes brute forcing them rather easy.

I was never concerned about short extension numbers and easy passwords 
until the need came up to connect to my * box from outside.

Thanks again!

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[asterisk-users] Restrict SIP registration to one ip address only?

[Remco Barendse]

Maybe a bit silly question, but why doesn't Asterisk accept if you set 
both a username&password as well as an ip address for a phone?

My fixed phones in my home all have a fixed ip address, but i also have 2 
Nokia GSM phones that can talk sip wich i would like to use from public 
wifi.

It's obvious that the more phones you have the more successful a brute 
force attack on the server will be, so i would only like to allow access 
to he 2 Nokia phones from "any" ip.

Why doesn't Asterisk allow both username&pass as well as setting an ip 
adress on a sip.extension?


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[asterisk-users] callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable

[Remco Barendse]

Hi list,

My caller ID is not working anymore on my TDM11B (TDM400P) cards and i get 
this error message on the asterisk console:

== Starting post polarity CID detection on channel 4
 -- Starting simple switch on 'Zap/4-1'
[Jul  8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf: 
Couldn't detect start-character. CID parsing might be unreliable

A long time ago my CallerID used to work with the same settings. I don't 
really need CallerID but it would be nice to have it working. I am located 
in The Netherlands.

Any suggestions?

This is in my /etc/zaptel.conf :
fxoks=1
fxsks=4 
loadzone=nl 
defaultzone=nl 
This is in my /etc/asterisk/zapata.conf :
echocancel=yes
echocancelwhenbridged=yes
echotraining=400

callerid=202
signalling=fxo_ks
group=1
context=intern-all
channel=>1

signalling=fxs_ks
immediate=yes 
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no
callwaiting=no 
callwaitingcallerid=no
adsi=no 
group=2 
context=inbound-analog
channel=>4


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Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

[Remco Barendse]

On Mon, 7 Jul 2008, Matt Gibson wrote:

> I think there is an issue with the screen refresh, mine also displays
> "searching..." unless I reboot the phone, and leave wifi on when it boots
> up, at this point it says "internet calling: available" .. but, it works
> either way.

or maybe i am using an old voipwm6.cab or sip config?  I also seem to have 
the problem that sound is only coming from the speaker on the back, not the ear 
speaker.

>
> As for prepending a 9, that's something your Asterisk installation is doing
> (or has been setup in the voip software to do)

I guess this has to do with a dialplan in the phone, googling showed me 
some posts about people modifying it. There is no 9 prefix in my asterisk 
dial plan.

Thanks!
Remco



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[asterisk-users] CallerID in The Netherlands with TDM11B

[Remco Barendse]

For quite a long time already my CallerID stopped working (maybe even when 
i upgraded from Asterisk 1.2 to Asterisk 1.4). I am using a TDM400P card 
(in TDM11B config) with one FXO and one FXS port.

Tried googling for some more recent examples of Asterisk config files for 
use in The Netherlands, but without success.

Anybody here that has this working?

When a call comes in i see this on the asterisk console :

   == Starting post polarity CID detection on channel 4
 -- Starting simple switch on 'Zap/4-1'
[Jul  8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf: 
Couldn't detect start-character. CID parsing might be unreliable


This is in my /etc/zaptel.conf :
fxoks=1
fxsks=4
loadzone=nl
defaultzone=nl

This is in my /etc/asterisk/zapata.conf :
echocancel=yes
echocancelwhenbridged=yes
echotraining=400

callerid=202
signalling=fxo_ks
group=1
context=intern-all
channel=>1

signalling=fxs_ks
immediate=yes
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
adsi=no
group=2
context=inbound-analog
channel=>4

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Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

[Remco Barendse]

Hi Matt!!

Thanks for that. When i use the same config, it looks like my Asterisk 
1.4.21.1 really expects the md5secret because i get this :
[Jul  7 11:11:21] NOTICE[17678]: chan_sip.c:15236 handle_request_register: 
Registration from '' failed for 
'10.10.250.252' - Wrong password

When i uncomment the md5secret, the phone seems to register but the 
display of the phone keeps on displaying 'Searching.' at the place of 
internet call but i can call my local voicemail.

When i try to call any other number i can see that the phone is dialling a 
9 before the number i want to dial.

Weird..


On Sun, 6 Jul 2008, Matt Gibson wrote:

> Hi Remco,
>
> Here's my SIP config..
>
> [8902]
> type=friend
> secret=xxx
> record_out=Adhoc
> record_in=Adhoc
> qualify=no
> port=5060
> pickupgroup=
> nat=no
> md5secret=xxx
> [EMAIL PROTECTED]
> host=dynamic
> dtmfmode=auto
> dial=SIP/8902
> context=from-internal
> canreinvite=no
> callgroup=
> callerid=device <8902>
> accountcode=mobile
> call-limit=50
>
> As for internet calling, once you have the feature installed, there is a new
> option available in the connection settings "use internet dialing when
> available". I have this turned on, and calls route over wifi/voip when I am
> registered instead of the cell network. Hth!
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barendse
> Sent: Sunday, July 06, 2008 8:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?
>
> On Thu, 3 Jul 2008, Matt Gibson wrote:
>>
>>
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
>> ws-mobile-6x-for-free-voip-calls-using-asterisk/
>
> Thanks for the link!
>
> I installed and configured the phone according to the above link.
>
> It only seems to work partly though. I sometimes managed to get tmy TyTN
> II to register with asterisk.
>
> Incoming calls only work sporadically, only when the phone registers as
> peer (which it doesn't always do). And even when inbound calls work, i
> never have the option to select Internet Calling when making a call.
>
> The phone does state "Internet Calling: Available"
>
> Maybe my sip.conf is wrong?  This is what i have now :
>
> [123]   ; HTC TyTN II
> disallow=all
> allow=alaw
> type=friend
> username=123
> secret=12345678
> host=dynamic
> insecure=port,invite
> callerid="HTC TyTN II" <123>
> context=intern-all
> nat=yes
> qualify=yes
> canreinvite=no
> dtmfmode=rfc2833
> notransfer=yes
>
>
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Re: [asterisk-users] Windows Mobile 6 IAX/SIP client?

[Remco Barendse]

On Thu, 3 Jul 2008, Matt Gibson wrote:
>
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
> ws-mobile-6x-for-free-voip-calls-using-asterisk/

Thanks for the link!

I installed and configured the phone according to the above link.

It only seems to work partly though. I sometimes managed to get tmy TyTN 
II to register with asterisk.

Incoming calls only work sporadically, only when the phone registers as 
peer (which it doesn't always do). And even when inbound calls work, i 
never have the option to select Internet Calling when making a call.

The phone does state "Internet Calling: Available"

Maybe my sip.conf is wrong?  This is what i have now :

[123]   ; HTC TyTN II
disallow=all
allow=alaw
type=friend
username=123
secret=12345678
host=dynamic
insecure=port,invite
callerid="HTC TyTN II" <123>
context=intern-all
nat=yes
qualify=yes
canreinvite=no
dtmfmode=rfc2833
notransfer=yes


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[asterisk-users] Windows Mobile 6 IAX/SIP client?

[Remco Barendse]

I just bought a HTC TyTn II phone, but unfortunately it doesn't even have 
a SIP client in it.

I tried the wiki searching for a SIP or IAX client but only found some 
PocketPC stuff (Windows Mobile 2003).

Does anyone know of a good quality SIP or IAX softphone that will run on 
Windows Mobile 6?

I only have a data subscription, no voice so the quality should be 
sufficient to be used constantly.

Thanks!!

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Re: [asterisk-users] Major problem with 1.4.21 asterisk

[Remco Barendse]

I think "the other guy" would be. me ?

Unfortunately i am also running my asterisk on a production environment 
where people start screaming the moment "it doesn't work"

I have 1.4.21 running at 3 locations in a home environment, simple TD400 
cards with analog ports and no problems.

My problem was with outgoing iax calls and the PRI going dead not 
accepting inbound calls.

If there is any interest i can try and test it this weekend, should i use 
the version with or without the mentioned patch?

Would anyone have some cripts to hammer the PRI from my home box? (Using 
an IAX termination provider)

Remco


On Thu, 26 Jun 2008, Michael J. Liberatore wrote:

> Hopefully the other guy with the problem can test it because this is a
> production server and the client is already upset about the problems
> this caused for a day or two till I realized what the issue is so I cant
> risk it.   Maybe I can off hours if he cant though.
>
> Mike
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
> Lesher
> Sent: Wednesday, June 25, 2008 9:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk
>
> On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote:
>> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having
>> major iax2 problems.  All of a sudden calls wouldnt come in on the
>> iax2 DID, and we couldnt make calls out even though everything looked
> ok.
>> Also there was usually a hung iax2 channel when this happened.
>> Stopping asterisk also wouldnt work, i would do a "Stop now" and it
>> would just go back to the cli prompt.  I would do a ? and it wouldnt
>> work.  I would have to kill asterisk via ps and then restart it via
>> init.d and then
>> iax2 would start working again for a short while (maybe a few hours)
>>
>> I reinstalled 1.4.19 and the problems went away.  There appears to be
>> a major bug in 1.4.21 but i am not sure.
>
> Please try the patch in bug number 12903:
> http://bugs.digium.com/view.php?id=12903
>
> --
> Tilghman
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[asterisk-users] Asterisk 1.4.21 stalls?

[Remco Barendse]

Ip upgraded yesterday from Asterisk 1.4.20.1 to 1.4.21

The update seems to work ok, when asterisk is started all is fine.

However after some time it is not possible to call anymore, my Snom 
display simply shows Not available and incoming calls from the PRI fail, 
like the PRI is not connected.

Reverting back to 1.4.20.1 solves the problem.

I tried re-installing and re-compiling 1.4.21 and all the modules several 
times, didn't help.

Anyone else seeing this?

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Re: [asterisk-users] Dialing patterns and "GSM" format numbers

[Remco Barendse]

On Fri, 14 Mar 2008, Adrian Merwood wrote:

> In my asterisk (Trixbox) server I would like to be able to dial
> numbers from my address book using HUD or the SIP client on my 3G
> phone using numbers in this format.
>
> On asterisk I would like to strip of the + and replace it with an
> international dialing prefix.
> Secondly (in the future) I would like to strip off certain country
> codes and replace them with a local dialing prefix.
>
> Can anyone help me figure ths out?


I got this workingm, thanks to a solution posted on this list but i didn't 
find a way how to implement this in FreePBX (Trixbox).

This is how i did it in my normal asterisk installation :

Include this context as first in your outbound pattern :

[intern-all]
include => prefix


the context that actually fixes the prefix is this :
[prefix]
exten = _+31.,1,Goto(0${EXTEN:3},1); Change +31 to 0
exten = _+.,1,Goto(00${EXTEN:1},1) ; Change + to 00


The tricky part to implement this properly in FreePBX is dissect the 
dialled telephone number to propely route calls through the appropriate 
provider. In my case i use the same provider for local calls as for 
international calls, but those that have free local calls do not want 
every call to a local number dialled out through their international 
provider. For me it's all the same :)

If you find a solution to properly do it in FreePBX, please post it to the 
list :))


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[asterisk-users] SIP Bad request protocol Packet on Asterisk 1.4.18

[Remco Barendse]

Hi all!!

I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk 
1.4.18. Both are home PBX's and both boxes register to a SIP DID at
exactly same provider. One box runs without errors on the console, the 
other box keeps repeating :

[Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705 
determine_firstline_parts: Bad request protocol Packet

When i set debug on, it seems to come from that SIP DID.
<--- SIP read from 82.101.62.99:5060 --->
Cirpack KeepAlive Packet
<->
[Feb 11 23:37:59] WARNING[11292]: chan_sip.c:6705 
determine_firstline_parts: Bad request protocol Packet
--- (1 headers 0 lines) ---

What i don't understand is why i get this message on one box only?

Ideas anyone?

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Re: [asterisk-users] GSM Gateway behind SIP ATA?

[Remco Barendse]

On Fri, 4 Jan 2008, EdPimentl wrote:

> Have you looked into
> http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
> -E

Yes i did, looks like an excellent product with many, many features and 
of outstanding quality.

However, given the cost of that unit i would have to be calling mobile 
phones 24 hours per day for at least the next 10 years of my life to 
earn the investment back, so definatively economically unviable.

But thanks for the tip :)

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Re: [asterisk-users] GSM Gateway behind SIP ATA?

[Remco Barendse]

>
> You can use the D option with the Dial command.
> Something like this should work:
> exten => _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})


It worked

Here is how i did it in FreePBX :

1) Setup a SIP extension for the ATA device, in my case i give it 
extension number 298. Edit the extension after creating it set DISALLOW to 
all and set ALLOW to alaw to make sure DTMF sending will work.

2) Create a custom trunk, and set as Custom Dial String :
Local/[EMAIL PROTECTED]

3) add to extensions_custom.conf :
[custom-gsmvoip-out]
exten => _.,1,Dial(SIP/298,,D(ww0${EXTEN}))

Note that i put a leading zero there, because for my fallback outbound 
routes i needed to strip the leading zero so i added it again here.

4) Insert the custom trunk in outbound routes

That's it

Hope this will save somebody else 2 days of frustration :)))

Cheers!

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Re: [asterisk-users] GSM Gateway behind SIP ATA?

[Remco Barendse]

On Thu, 3 Jan 2008, Benchev wrote:

> Basically Grandstream HT286 is a single port FXS ATA.
> In order to interconnect GSM gateway one would need FXO.
> Are you sure it gives you "new" dialing tone or this is the * itself
> you hear?

Yes, i am positive that i get a new dialtone from the GSM Gateway.

If i dial DTMF codes from a SIP phone connected to Asterisk, i can see the 
digits appear in the display of the GSM Gateway. But it is a bit 
incovenient to call an internal extension, wait for the dialtone and then 
punch in all the numbers of the cell phone i need to call.

I would prefer Asterisk to decide where / how to route the call and send 
the DTMF inband to the ATA device.

Thanks!!


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[asterisk-users] GSM Gateway behind SIP ATA?

[Remco Barendse]

I have an analog GSM Gateway that is connected to a normal SIP ATA device.

Basically what it does is this : when you call the extension nr. of the 
SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) 
dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia 
a Grandstream HT286.

I would like to use the GSM Gateway to route my outbound cellular calls, 
how do i do this in Asterisk? Basically Asterisk should dial the extension 
number and then send required number as DTMF tones to the Gateway through 
the ATA.

I am using FreePBX, which allows me to create a custom trunk for the 
outgoing calls. Hope this could work :)

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[asterisk-users] Asterisk 1.2.26 badly broken?

[Remco Barendse]

After upgrading from 1.2.25 to 1.2.26 i noticed that IAX -> IAX calls 
always result in Asterisk just exiting without any message.

Asterisk also seems to die when using a TDM400 with 2 FXO modules, placing 
2 outgoing calls on both lines as Zap/g2 and then trying to make a 3rd 
call.

Went back to 1.2.25 for the moment

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Remco Barendse]

> I wonder if there are any major obstacles for upgrading.


Just tried an in-place upgrade on my home box :

make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so 
res_config_mysql.so; do /usr/bin/install -c -m 755 $x 
/usr/lib/asterisk/modules ; done
/usr/bin/install: cannot stat `app_addon_sql_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `cdr_addon_mysql.so': No such file or 
directory
/usr/bin/install: cannot stat `res_config_mysql.so': No such file or 
directory
make: *** [install] Error 1


And the asterisk console is flooded with these errors :

[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet
[Dec 21 10:10:58] WARNING[22897]: chan_sip.c:6707 
determine_firstline_parts: Bad request protocol Packet

So for the next time to come i'll turn back to 1.2 :)

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Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

[Remco Barendse]

> I wonder if there are any major obstacles for upgrading.

My reasons for not moving to 1.4 :
- fear of possible instability problems, my 1.2 servers are rock solid
- fear of goofing up with the new way you have to configure asterisk
   at install time (tell it which modules to build or not build)
- no real new functionality i really, REALLY need

One feature that would immediately draw attention and would greatly 
enhance upgrade enthusiasm for a new release would be better fax support.

chan_mobile looks nice, would be nifty to be able to use gsm phones, i 
will probably look into that

Just my $0.02 :)

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Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config

[Remco Barendse]

On Fri, 26 Oct 2007, Benny Amorsen wrote:

>>>>>> "RB" == Remco Barendse <[EMAIL PROTECTED]> writes:
>
> RB> Hi list! Is anyone using the Kirk IP600/3 with SIP firmware
> RB> connected to Asterisk?
>
> Yes.
>
> RB> If anyone would be willing to share the dump of their IP600 config
> RB> file, i would really appreciate it.
>
> Sorry I'm not at work right now. If I get time later, I will.

Hi Benny!

Did you manage to make a dump of a working configuration from the IP600/3?

Would be really useful, can't seem to get it to work properly.

Thanks!
Remco

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[asterisk-users] Kirk IP600/3 Wireless Server SIP config

[Remco Barendse]

Hi list!

Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk?

Any experiences / caveats?

If anyone would be willing to share the dump of their IP600 config file, 
i would really appreciate it.

Is there anything special i should put in my asterisk config?

Thanks !!!
Remco

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Re: [asterisk-users] Nokia cell connected to Asterisk

[Remco Barendse]

Has anyone ever tried using a Nokia phone with SIP client as channel for 
Asterisk?  I mean i would like to receive calls to the mobile on 
asterisk and use the Nokia phone to place calls to cell destinations.

I have enough Nokia E60's to do that and it would circumvent the need for 
chan_bluetooth or something similar!! :)


On Mon, 20 Aug 2007, Steve Totaro wrote:

> Well chan_bluetooth is really amazing (especially if your phone does not
> support SIP).
>
> You connect your phone via bluetooth to your asterisk box and it becomes
> a channel type.  You can use it as an extension(FXS) or a phone line
> (FXO).  I believe you can send and receive SMS through the
> phone/Asterisk as well.
>
> Chan_bluetooth README is in the asterisk-addons trunk and gives you
> basic instruction on setting it up.
>
> You get several added pieces of functionality with this setup.  SMS send
> and receive through your phone using Asterisk?, FXO failover or LCR, FXS
> where your cell phone becomes an extension.
>
> Thanks,
> Steve
>
> Jonathan GF wrote:
>> Thanks Steve and Mitcheloc,
>>
>> in fact i was think in something more "obsolet" like connect via
>> serial/usb cable the cell to the asterisk box. Never thought in the
>> SIP stack of new Nokia's but i will start looking for info about this.
>> If you [Steve] know of a good written material of interest please let
>> me know.
>>
>> Probably Mitcheloc is right too, there are a lot of manners to achieve
>> this and the problem is mine that i don't know how to search what i
>> want. Anyway, thank you for your inputs. Any others will be welcomed,
>> for sure.
>>
>> Regards,
>>
>> Jonathan GF
>>
>>
>>
>> On 8/20/07, *mitcheloc* <[EMAIL PROTECTED]
>> > wrote:
>>
>> Jonathon,
>>
>> Are you talking about using the built in SIP client on some Nokia
>> phones? I'm using an E90 with Asterisk and it works very well. I used
>> Google for help and it returned plenty of results.
>>
>> Cheers,
>> Mitchel
>>
>> On 8/19/07, Steve Totaro <[EMAIL PROTECTED]
>> > wrote:
>>> If it is bluetooth and you don't mind running Asterisk 1.4
>> trunk, you should look at chan_mobile.
>>>
>>> Thanks,
>>> Steve Totaro
>>>
>>> 
>>>
>>> From: [EMAIL PROTECTED]
>>  on behalf of
>> Jonathan GF
>>> Sent: Sun 8/19/2007 6:26 PM
>>> To: asterisk-users@lists.digium.com
>> 
>>> Subject: [asterisk-users] Nokia cell connected to Asterisk
>>>
>>>
>>> Hi folks,
>>>
>>> i've been looking for in many sources but i cannot see clear if
>> the options i'm chasing is feasible with Asterisk. I understand
>> that should be.
>>>
>>> I would like to connect a nokia cell to Asterisk but i don't
>> know how exactly.
>>>
>>> Any ideas, inputs, docs or refs will be welcomed.
>>>
>>> Thanks in advance.
>>>
>>> Jonathan GF
>>>
>>>
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>>>
>>
>>
>> --
>> 
>> Mitchel Constantin
>> Snap - A desktop user interface for Asterisk
>> www.snapanumber.com 
>>
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>> 
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[asterisk-users] Force asterisk to re-resolve dns names?

[Remco Barendse]

Is there really no way to have asterisk re-resolve domain names from iax 
or sip providers if this failed or timed out the first time?

When asterisk boots on every box i have asterisk is t impatient 
trrying to resolve the domain names for a first time. This results in 
asterisk thinking the provider is unreachable and only trying again in one 
week or so.

This results (depending on the dial plan) on either not being able to make 
calls or to see all calls flow out via (extremely expensive) PSTN.

This 'feature' of asterisk really pisses me off, why can't it just 
re-resolve the few host names again within a reasonable amount of time???

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Re: [asterisk-users] VPN on Asterisk

[Remco Barendse]

> Hi,
>
> Greetings to All,
>
> Im looking for some help on configuring VPN on the Asterisk PBX that I
> have hosted in US. Im currently in Middle East and as everyone knows
> some countries here has taboo to VOIP. Im not able to get phy phones
> registered to my PBX as they are blocking SIP and IAX2. Hence im
> looking for a VPN solution.

Slightly offtopic, but I would choose a VPN solution that can do webvpn 
(connect to port 80), i just came back from holiday and several hotels had 
VOIP *and* VPN ports for PPTP blocked in their internet, to prevent people 
from calling over their internet connection, clogging up their (pretty 
poor) connection. With webvpn you can connect to port 80 and circumvent 
such trouble.

I tried finding an easy HOWTO for OpenVPN, on a CentOS box, this is not 
easy at all.


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[asterisk-users] Recent zaptel versions break CLIP?

[Remco Barendse]

Hi!

Is it just me or do the last 2 or 3 versions of the zaptel-1.2 branch seem 
to break cli? Often not the full number is displayed, or only 2 or 3 
digits?


I am in The Netherlands, and have had this in my zapata.conf (which used 
to work flawlessly) :

signalling=fxs_ks
immediate=yes
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no


I installed some new boxes with newer zaptel cards recently, but same 
problem.


Thanks for any hints / tips!
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Re: [asterisk-users] [OT] Nokia E60 firmware update break SIP

[Remco Barendse]

On Mon, 16 Apr 2007, Martin Joseph wrote:

Just a warning for you all that are using Nokia series E phones for SIP 
function.


I updated my phones firmware today using the Nokia Updater,  and now the SIP 
functionality, which previously worked pretty well is completely broken.


The phone no longer registers with asterisk, although it displays the little 
icon as though it has, and it doesn't even seem to try to pass calls to 
asterisk...


So,  I would avoid 3.06330904 20-11-06 RM-49


Where did you find this version?  My Nokia updater only offers an update 
to 2.0something  (my phone had 1.0something)


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Re: [asterisk-users] error in FreePBX

[Remco Barendse]

On Thu, 29 Mar 2007, Carlos Jerónimo wrote:


Ive installed asterisk and freepbx. Through the interface ive
configured 2 extensions, 6000 and 6001.
My problem is that when i try to call from extension 6000 to 6001, i
hear this msg "Im-sorry&an-error-has-occured" and the call is
terminated.
As expected if i call to another number i get an error.
i thought the problem might been related with the NAT but if checked
and changed some NAT configuration parameters, it didnt worked aswell.
As this ever happened to anyone before? Any hints are very appreciated.

Thank you very much


I have the same problem, it seems to occur when an extension is busy here.

All my extensions are on local lan with phones having ip addresses in a 
private range without NAT or anything so that is not the problem.


Sounds like an error in the dial pan FreePBX generated.___
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Re: [asterisk-users] TDM02B not working

[Remco Barendse]

Aarghh nevermind, my bad

A stock TDM02B comes with modules installed in slot 3 and 4, not 1 and 2.

For whoever might have the same problem and finds this post change below 
to read:

fxsks=3-4
channel=>3-4

On Sun, 11 Feb 2007, Remco Barendse wrote:

I am trying to reconfigure an asterisk box that was using an HFC-S card with 
bristuff but is now using 2 analog lines therefore I want to use the TDM02B 
to connect to two POTS lines. The TDM02B has 2 red modules.


I have this in /etc/zaptel.conf
loadzone=nl
defaultzone=nl
fxsks=1-2

I have /etc/asterisk/zapata.conf
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400

immediate=yes
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
adsi=no
context=inbound-analog
channel=>1-2

The output of ztcfg -v is :
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.

so far so good, but when i try asterick -c :
 == Parsing '/etc/asterisk/zapata.conf': Found
Feb 11 16:48:15 WARNING[19083]: chan_zap.c:1072 zt_open: Unable to specify 
channel 1: No such device
Feb 11 16:48:15 ERROR[19083]: chan_zap.c:7034 mkintf: Unable to open channel 
1: No such device

here = 0, tmp->channel = 1, channel = 1
Feb 11 16:48:15 ERROR[19083]: chan_zap.c:10462 setup_zap: Unable to register 
channel '1-2'
Feb 11 16:48:15 WARNING[19083]: loader.c:414 __load_resource: chan_zap.so: 
load_module failed, returning -1
Feb 11 16:48:15 WARNING[19083]: loader.c:554 load_modules: Loading module 
chan_zap.so failed!


Why doesn't asterisk find the channel that ztcfg does see?
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[asterisk-users] TDM02B not working

[Remco Barendse]

I am trying to reconfigure an asterisk box that was using an HFC-S card 
with bristuff but is now using 2 analog lines therefore I want to use the 
TDM02B to connect to two POTS lines. The TDM02B has 2 red modules.


I have this in /etc/zaptel.conf
loadzone=nl
defaultzone=nl
fxsks=1-2

I have /etc/asterisk/zapata.conf
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400

immediate=yes
usecallerid=yes
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
adsi=no
context=inbound-analog
channel=>1-2

The output of ztcfg -v is :
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.

so far so good, but when i try asterick -c :
  == Parsing '/etc/asterisk/zapata.conf': Found
Feb 11 16:48:15 WARNING[19083]: chan_zap.c:1072 zt_open: Unable to specify 
channel 1: No such device
Feb 11 16:48:15 ERROR[19083]: chan_zap.c:7034 mkintf: Unable to open 
channel 1: No such device

here = 0, tmp->channel = 1, channel = 1
Feb 11 16:48:15 ERROR[19083]: chan_zap.c:10462 setup_zap: Unable to 
register channel '1-2'
Feb 11 16:48:15 WARNING[19083]: loader.c:414 __load_resource: chan_zap.so: 
load_module failed, returning -1
Feb 11 16:48:15 WARNING[19083]: loader.c:554 load_modules: Loading module 
chan_zap.so failed!


Why doesn't asterisk find the channel that ztcfg does see?
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Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card

[Remco Barendse]

On Sun, 11 Feb 2007, Leo Ann Boon wrote:


Matt wrote:


 I guess the question is... is it even possible to have a real-time VoIP
 card running on PCIe?  Or with 1,000 Interrupts a second.. does it simply
 need to have its own IRQ?

Have you tried the Sangoma PCIe cards?

APIC is supposed to fixed the PCI IRQ problem. AFAIK, APIC is not a virtual 
interrupt. It requires an additional interrupt controller to deal with the 
additional interrupt lines. The BIOS cannot see it because it's still stuck 
with the 8086 15-interrupt mindset. When you run a modern OS like Windows XP 
and Linux, the OS can will make the CPU aware of the additional interrupts 
from the secondary interrupt controllers. At the BIOS level, you'll see 
'shared' interrupts for APIC system because the mobo designer need to cascade 
the new interrupt controller to the standard controller. Otherwise, the 
interrupts from the secondary controller will not be available to real-mode 
applications.


If the above would work like it was meant to be why do many cards still 
have irq problems?  In zttest i only get 99.987793% scores, not higher, 
not lower only a very rare 100%.


My cheap ass Asus A78VX-X board scores considerably better then this 
expensive Dell machine with the same Digium card and software installed.


I have apic enabled, disabling apic from the kernel did not help to 
improve things


My simple conclusion from the above is that the Dell hardware sucks.

result from zttest :
--- Results after 198 passes ---
Best: 100.00 -- Worst: 99.987793 -- Average: 99.988498

[EMAIL PROTECTED] zaptel-1.2.13]# cat /proc/interrupts
   CPU0
  0:   82179609IO-APIC-edge  timer
  2:  0  XT-PIC  cascade
  8:  1IO-APIC-edge  rtc
 14: 738454IO-APIC-edge  ide0
 74:   82142210   IO-APIC-level  wct2xxp
201: 745906   IO-APIC-level  megaraid
209: 137704   IO-APIC-level  eth0
217: 177366   IO-APIC-level  eth1
NMI:  0
LOC:   82178536
ERR:  0
MIS:  0
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Re: [asterisk-users] Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card

[Remco Barendse]

On Sat, 10 Feb 2007, Andres wrote:




 try booting with APIC and ACPI disabled?

Thats right.  I have never seen a shared IRQ with Dell servers using APIC.  A 
RHL ES3 by default enables APIC so I have never even had to fiddle around 
with it.


Ofcourse you don't. But simply because APIC makes you believe that IRQ's 
are not shared, that doesn't necessarily mean they really aren't1!

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Re: [asterisk-users] Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card

[Remco Barendse]

On Sat, 10 Feb 2007, Matt wrote:


Hi folks.. just a few weeks ago I wrote this to someone else:

We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:

* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put ETH0 on a seperate IRQ from the Digium card

And everything's fine.  Dell's do NOT have to share IRQs... go into your
BIOS and change them


And this is still true.  However, we recently got a 2950 to use as a VoIP
server with a digium 4 port TDM2400 (I believe) analog card.   Well wouldn't
you know the Dell BIOS is showing NIC1 AND NIC2 AND Digium Card sharing
the same IRQ.  No matter what I change one of them two, the other two
follow.   I've tried moving the Digium card to the other PCI slot and the
IRQ problem still exists.  I talked to Dell technical support and they
said "oh all our new machines share IRQs like that, the way you are trying
to do it is archaic".  What?!?!   The Dell tech guy kept saying that I can
define an IRQ in Linux, and I kept telling him that I need two unique (not
virtual) IRQs.. one for the NIC and one for the Digium card.  He said "yeah
we've had other calls about Digium cards in these servers not working".
ARG!Now I'm in a real quandry.


Lol, see my replies to the thread.  This crappy Dell shit always shares 
irq's.


For my next servers I'll be ordering Arima mainboards I think and assemble 
the things myself again.


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Re: [asterisk-users] Dell Servers

[Remco Barendse]

On Mon, 5 Feb 2007, Matt wrote:


We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:

* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put ETH0 on a seperate IRQ from the Digium card

And everything's fine.  Dell's do NOT have to share IRQs... go into your
BIOS and change them.


I disagree, when you are not using APIC on a Dell 2850 you can assign an 
IRQ to each PCI slot but each pci slot is linked to an internal device. No 
matter what you do, if you assign for example IRQ 10 to PCI slot 1 also 
the SCSI controller will use that IRQ.


With just 3 pci slots this just totally sucks, this is totally 
unnecessary and shows poor design.


I do not experience any lockups, just poor IRQ hits in zttest and dropped 
calls.


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Re: [asterisk-users] Dell Servers

[Remco Barendse]

On Sat, 3 Feb 2007, Gordon Henderson wrote:


On Fri, 2 Feb 2007, Remco Barendse wrote:


 Would you be willing to share your blacklist for the kernel modules?


Have you considered compiling a custom kernel for your hardware rather than 
not loading modules? It's something I've always done from day 1 with Linux 
(some 10 years back now!) That way I get exactly what I need and no more, and 
the only modules I have are the wctdm & zap ones.


It's not everyones cup of tea though - especially when using a distributing 
that you need to keep "standard" for external support purposes.


Thanks! It should give me some pointers. Indeed compiling kernels is not 
my cup of tea. Everytime I try it on some stupid box i couldn't care less 
about it works flawlessly but everytime I try it on a production box 
(preferably remote) the box will not boot after :)


So for the moment I'll stick with stock redhat.
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Re: [asterisk-users] Dell Servers

[Remco Barendse]

On Thu, 1 Feb 2007, Christophorus Laube wrote:


We have a 2850 in a productive environment with a BNE1 performing well
(OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu
Edgy). You only have to blacklist some hotplug kernel modules and yes, we do
have very long pings (1 ping per week with a check rate of 10min per SNMP).
But that does happen very rare and I never noticed any dropped calls or bad
audio quality. The 2850 is running on SCSI, the 2950 on an SAS RAID.
In general I like the Dell machines, also with asterisk on them. The only
thing is that Openmanage ist quite bad to install but that's nothing asterisk
specific but linux related.
Does that help?


Would you be willing to share your blacklist for the kernel modules?

Thanks!!

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Re: [asterisk-users] Dell Servers

[Remco Barendse]

On Thu, 1 Feb 2007, Eric Rousse wrote:


Hi,

I was planning on getting a Dell PowerEdge 2950 for our new Asterisk 
configuration.
But while searching for documentation about it and/or reported issues, I 
found this:


http://www.voip-info.org/wiki/view/Asterisk+hardware
WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, 
which has been known to cause random locksup - if you plan on using a Dell 
server, disable the onboard controller and purchase an addon ethernet card.


Does anyone has real experience ?


I bought a Dell 2850 as a pbx server and it just sucks IMHO

The stupid thing has only 3 pci slots and even with only 3 pci slots Dell 
managed to have a shared irq on every slot, 1 for the scsi controller and 
one for each nic


The result of this 'nice' piece of work is dreadfull irq hit/miss results 
in zttest, it barely meets the minimum requirement and i do get complaints 
of dropped calls on my pri


I need to pass some options to the kernel at boot time to improve things, 
without extra options the results from zttest were unacceptable


My spare pbx is a lowly Athlon XP 2600 with an Asus A7V8X-X mobo in it and 
it's scores with zttest are considerably better (but not full 100% hits)


I know that everybody on the list will now start recommending me to buy 
Sangoma hardware but firstly I hate compiling extra modules and it doesn't 
make it right that the Dell hardware just sucks


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Re: [asterisk-users] vzaphfc?

[Remco Barendse]

On Wed, 3 Jan 2007, Tzafrir Cohen wrote:


P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS
EXPERIMENTAL!)
..Jan  2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown
signalling method 'bri_cpe_ptmp'


our Asterisk is not bristuffed. And you don't expect to use ZapBRI,
anyway.

BTW: with 1.4 and latest 1.2, bristuffed zaptel could basicaly work with
the signalling  type pri_cpe/pri_net, though this is not well-tested and
may not perform as well as bristuffed asterisk/libpri.


Jan  2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must
be specified before any channels are.
Jan  2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so:
load_module failed, returning -1
Jan  2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module
chan_zap.so failed!


You have zaptel channels configured in your zapata.conf .

So I should leave both zaptel.conf and zapata.conf completely empty?


BTW: I guess that this is 1.2 and not 1.4, because 1.4 should not fail
just becasue chan_zap.so has failed to set up some channels.


This is Asterisk 1.2 indeed, i need the chan-sccp driver from Sergio 
Chersovani because it supports multiple phone registrations from 1 ip 
address and AFAIK that channel doesn't work on 1.4 yet. Is there a lot of 
difference in misdn support between 1.2 <-> 1.4?


I tried googling for example configs with misdn but couldn't find any.

Thanks!
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Re: [asterisk-users] vzaphfc?

[Remco Barendse]

On Fri, 29 Dec 2006, Julian J. M. wrote:


It's not necessary to recompile the kernel for mISDN support. Check
http://www.laimbock.com/asterisk/

Grab the mISDN source rpm, and build it.

$ wget 
http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm

$ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm

then check /usr/src/redhat/RPMS/i386/
You should have the kernel modules and userspace applications. Once
installed, I could enable chan_misdn in asterisk 1.4 without issue,
and it's working great in NT mode with ISDN phones. I haven't tested
asterisk 1.2, but there is no it shouldn't work as well.


I deleted all the bristuff modules i could find plus the old asterisk 
libs, compiled zaptel, libpri and asterisk from scratch but can't get it 
to work.


First I get errors about something I guess is missing from misdn, later 
errors about zaptel.


I'll just toss the HFC-S card and convert the ISDN line to analog.

These are the errors :
.mISDN_close: fid(14) isize(131072) inbuf(0x2a96ee6010) irp(0x2a96ee6010) 
iend(0x2a96ee6010)
Jan  2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: 
misdn.conf: "ports=(null)" (section: intern) invalid or out of range. 
Please edit your misdn.conf and then do a "misdn reload".
Jan  2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: 
misdn.conf: "ports=(null)" (section: first_extern) invalid or out of 
range. Please edit your misdn.conf and then do a "misdn reload".
Jan  2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: 
misdn.conf: "ports=(null)" (section: second_extern) invalid or out of 
range. Please edit your misdn.conf and then do a "misdn reload".

P[ 0] Got: 1 from get_ports
P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS 
EXPERIMENTAL!)
..Jan  2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown 
signalling method 'bri_cpe_ptmp'
Jan  2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must 
be specified before any channels are.
Jan  2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so: 
load_module failed, returning -1
Jan  2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module 
chan_zap.so failed!


Cheers!
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Re: [asterisk-users] vzaphfc?

[Remco Barendse]

On Fri, 29 Dec 2006, Julian J. M. wrote:


It's not necessary to recompile the kernel for mISDN support. Check
http://www.laimbock.com/asterisk/

Grab the mISDN source rpm, and build it.

$ wget 
http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm

$ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm

then check /usr/src/redhat/RPMS/i386/
You should have the kernel modules and userspace applications. Once
installed, I could enable chan_misdn in asterisk 1.4 without issue,
and it's working great in NT mode with ISDN phones. I haven't tested
asterisk 1.2, but there is no it shouldn't work as well.

Julian J. M.


Sounds great thanks for the pointer!!  I'll give it a try tonight

Cheers!
Remco
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[asterisk-users] asterisk doesn't know version of asterisk-addons?

[Remco Barendse]

Hi!

I noticed when upgrading asterisk that the latest version of asterisk is 
not recognizing the version of asterisk-addons properly.


When you clean out /usr/lib/asterisk/modules and then install 
zaptel-1.2.12 -> libpri-1.2.4 -> asterisk 1.2.14 -> asterisk-addons-1.2.5 
and then you compile and install asterisk *again* it complains that the 
modules of asterisk-addons are not built for this version of asterisk?


Weird eh?
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Re: [asterisk-users] vzaphfc?

[Remco Barendse]

On Thu, 28 Dec 2006, Michiel van Baak wrote:


When you found out stuff, specially how to make stuff with a
simple HFC-S card stable please let me know.
We are not deploying them cards anymore because we never get
it stable.
Real simple setups can be done with a FRITZ!PCI card, but I
really prefer the quadbri cards for ISDN2


I think I'll try misdn or vzaphfc, if it is too complicated or i'm not 
satisfied with the results I will simply hook up a good old A/B adapter, 
convert the ISDN to analog lines and throw in a Digium TDM card.


I've pretty much had it with ISDN2
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Re: [asterisk-users] vzaphfc?

[Remco Barendse]

On Thu, 28 Dec 2006, Gavin Hamill wrote:


On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote:


vzaphfc is not a complete replacement of bristuff. It replies on most of
it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI
driver for HFC-s-based PCI cards.


Further, if you're looking for 'something else' re: cheapo ISDN cards,
definately give Asterisk 1.4 and mISDN a look - no BRIStuff, no huge patches,
no wacky stuff.. all Asterisk-core support that worked really well in the
brief time I tested it.

The key difference is rather than generating 8000 interrupts per second, the
mISDN kernel driver (which itself can be thought of 'isdn4linux' version 2.0)
polls the card, leading to much lower system load, and no 'wanted 8 bytes,
read 7!' errors from dmesg.


Thanks for the tip, I'll have a look at it. The main reason for me to use 
bristuff is that i don't want to mess mess around downloading and 
compiling my own kernels. I am just running CentOS 4 boxes with stock 
CentOS 4 kernels. Everytime I was screwing around with making my own 
kernels sooner or later I got bitten by screwing up the installation of 
the kernel and the box wouldn't boot anymore. :)


On the wiki I found the manual from BeroNet which looks pretty 
straightforward but is for Asterisk 1.2


Any differences for Asterisk 1.4?

Thanks!!
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[asterisk-users] vzaphfc?

[Remco Barendse]

Hi list!

I'm totally fed up with bristuff (or it's instability with a simple HFC-S 
card), 2 out of 3 times when people try to call they get the information 
tone that the number is not connected.


I would like to try vzaphfc and I am looking for information on it.

From previous posts I found that the only place where the sources seem to 

be maintained and available is at the debian site which I found here :
http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/

But I couldn't find any place where I could download a tarball.

Is vzaphfc an inplace replacement of the zaptel of bristuff? Or something 
separate?


Thanks!!
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[asterisk-users] New installation CentOS 4 x86 or X86_64

[Remco Barendse]

Hi list!

I have to do a new bare metal installation of a box running Asterisk with 
bristuff or vzaphfc.


The box will be used as a really lightly loaded file server and pbx.

Any advise on which architecture I should use? The cpu is a 64 bit capable 
AMD (the box is running x86_64 now) but is still suffering from echo on 
the BRI lines.


Should I go with the normal x86 or the 64 bit x86_64 arch.?

Thanks!!
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[asterisk-users] Can zaptel freak out if you configure 2 trunks but use only one?

[Remco Barendse]

I am using Asterisk 1.2.13 with Zaptel 1.2.11, I used to have an old PBX 
connected to one port and the PRI connected to the other.


I'm having serious stability issues with Asterisk on a box that has been 
rock solid previously.


The old PBX died two months ago so one port on the TE210P is now unused 
but still configured. Also I'm afraid I have upgraded from Asterisk 
1.2.9.1 and the old zaptel version because of the security flaws.


I'm now puzzled why Asterisk is being unstable.

I do a nightly restart because Asterisk is extremely slow in trying to 
resolve failed dns lookups for providers.


Often Asterisk will keep running or restart properly but on the console I 
can see it restarting / restarting the B-channels really slow (normally to 
restart all B channels takes max 1 second) i can really see it restarting 
one channel in about one second, and some channels are skipped.


It also happens that Asterisk refuses to start at all.

A reboot seems the only solution to the problem.

Could it be that the configured (but unused) trunk is causing me problems, 
buffer overruns or anything similar?  Or is this an issue with more recent 
zaptel and asterisk configs?


Thanks all!!
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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Remco Barendse]

> BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that
> the only place from which you can download an up-to-date version
> nowadays is the Debian zaptel package:
> 
> http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/
> http://packages.debian.org/zaptel-source

Thanks!  I tried looking for some more info on the Debian pages about what 
vzaphfc exactly is, but couldn't find any documentation of it.

Is there any main page?

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RE: [asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?

[Remco Barendse]

On Mon, 23 Oct 2006, Andreas Sikkema wrote:

> Remco,
> 
> > Asterisk starts before the internet connection is up and dns 
> > is working.
> 
> 
> 
> > And then people say nightly asterisk restarts are not a good idea
> 
> 
> Why is your asterisk startup script running before networking has been 
> setup? Asterisk has the same networking dependencies as apache, so I 
> start it around the same time using the same priority as apache and as 
> far as I know networking should work at that time or not at all, not 
> somewhere in between.

It is not, asterisk is correctly started after networking services, 
however it seems that when the box is booting the dns is replying just a 
split second too late for the taste of asterisk and it seems that asterisk 
then marks the provider as unavailable.

* should never wait that long, the 'load' on the box to resolve maybe a 
handful of domains is nothing, even if you would be running a Pentium 1 
box, and this should not be any reason not to try again every few minutes 
or so.


 
> pebkac?

If your view is broad enough all computer / it related trouble could be 
traced back to that :)
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[asterisk-users] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?

[Remco Barendse]

After a reboot, asterisk is usually too much in a hurry to try and resolve 
my iax/sip providers.

Asterisk starts before the internet connection is up and dns is working.

Then asterisk just waits, and waits and waits and waits even longer before 
ever trying to revolve any voip provider again.

And all this time calls are flowing out through the very expensive PSTN.

And then people say nightly asterisk restarts are not a good idea
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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Remco Barendse]

On Sat, 21 Oct 2006, Michiel van Baak wrote:

> On 20:15, Sat 21 Oct 06, Tzafrir Cohen wrote:
> > Interesting. Latest bristuff chenges the default Zaptel echo canceller
> > to MG2 (which is also the recommendation of Digium now). 
> > 
> > 
> > BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that
> > the only place from which you can download an up-to-date version
> > nowadays is the Debian zaptel package:
> > 
> > http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/
> > http://packages.debian.org/zaptel-source
> 
> Tzafrir,
> 
> Are you in the position to get stuff into asterisk ?
> The patch you sent me is working great. I think we need this
> stuff into the normal asterisk.
> BRI is something a lot of asterisk users depend on, how odd
> this may sound to USA ppl.

Which patches are that vzaphfc? I'm interested too :)
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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Remco Barendse]

> Ah, you are using it for the hfc-pci cards. That's a valid
> reason ;)
> What's the latest BRISTUFF version that does work on x86_64
> for you ? One of our customers reported trouble with this
> bristuff on x86_64 as well and I dont have a _64 machine to
> test.

The last working release is the one that works with asterisk 1.0 still :(

I have not rebooted the box to try the latest version but I'm having all 
sorts of issues with 0.3.0-PRE-1s.

To name a few :
Often incoming calls are not working, the caller gets the information tone 
as if there is no ISDN device connected to the line. Retrying the call 
works but this is extremely annoying.

Once I encountered the same problem on an incoming IAX call. The call 
would fail as if the server is not available.

The florz patch (which used to make the 0.2 series of bristuff reliable 
and stable) will not work on x86_64. After applying the patch I cannot get 
zaphfc to compile. In the past after applying the florz patch it used 
to be necessary to modify the Makefile and point it to the directory 
where the current kernel resides, lately even that doesn't work 
anymore.

I assume the first two problems are timing related, the florz patch would 
make bristuff behave better, without the patch I also have increased echo 
problems. (On a box without any load)
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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Remco Barendse]

> > And ofcourse half of the modules from this release do not build on a 
> > x86_64 box :(
> 
> Could you please be more spesific?
> 
> What distribution? What kernel? What errors?

Found the problem, it was fairly limited, where all the digium 
stuff finds the kernel automagically, the bristuff things still need a 
symlink in place. Then things seem to build.

Still have to reboot the box to see if it works though
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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Remco Barendse]

On Sat, 21 Oct 2006, Michiel van Baak wrote:

> On 09:39, Sat 21 Oct 06, Remco Barendse wrote:
> > And ofcourse half of the modules from this release do not build on a 
> > x86_64 box :(
> 
> What is it that you use in bristuffed that is not in plain
> asterisk ? I found myself battling with bristuff all the
> time as well and switched to plain asterisk at home and that
> works fine now for months.


Uhmm nothing, but the alternative would be mISDN I think, I am running 
CentOS and default kernels. I don't want to go down the path of compiling 
my own kernels so that is a show stopper for me.

I didn't look much further than compiling your own kernel in the misdn 
instructions but it looked too complicated to me. :) 
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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Remco Barendse]

On Fri, 20 Oct 2006, Michiel van Baak wrote:

> On 02:39, Fri 20 Oct 06, Tzafrir Cohen wrote:
> > On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote:
> > > On 23:04, Thu 19 Oct 06, Vidar wrote:
> > > > Bristuff has been updated;
> > > > 
> > > > http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
> > > 
> > > Thanks for the information.
> > > 
> > > It's a shame we need to read this here and not see it on
> > > their website.
> > 
> > Also note that the changelog entry for 0.3.0-PRE-1u is missing from the
> > CHANGES file. Nevertheless, that is a version for Asterisk 1.2.13 ,
> > Zaptel 1.2.10 and libpri 1.2.4 .
> 
> Also note that the changelog mentioned -1t and that that
> file is also available on FTP.
> I think junghann.net needs a webmaster ;-)

And ofcourse half of the modules from this release do not build on a 
x86_64 box :(

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Re: [asterisk-users] Re: Centos kernel 34 vs. 42? [was: asterisk-users Digest, Vol 27, Issue 72]

[Remco Barendse]

On Sun, 15 Oct 2006, Les Bell wrote:

> Cutting to the chase: I'm not aware of any audio problems, but our system
> doesn't get heavy use (only two lines and eight phones).

OK, thanks for the reply.

The anouncement at trixbox.org is not very clear on this. There is 
reference to 'distorted voice prompts' but not about general voice 
quality.

>From trixbox.org :
quote
There are some strange audio problems with the 42 kernel. This is most 
apparent with vmware. When the 42 kernel is used the audio prompts are 
jittery and broken. This was not a problem with the 34 kernel.

I am making the 34 kernel the standard for trixbox until further notice. 
If somebody has a good reason to move to a higher release kernel please 
post to the forum. But for now this should resolve all the problems with 
Zaptel. This update has a new yum configuration file that will keep yum 
from updating the kernel. If you want to update the kernel you can do it 
manually
unquote

I just went back to kernel 35 just in case.


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