RE: [asterisk-users] loosening voicemail file permissions formsg????.txt and msg????.wav

[Guerid Salim]

HI,

 

I’ve had the same problem,
and have done the same modification in the app_voicemail.c but without success.

It’s a bug related
in http://groups.google.ch/group/linux.debian.bugs.dist/browse_thread/thread/e4352f488af374bf/c667bae7bdc59473?lnk=st&q=app_voicemail.c+Bug%23330132&rnum=1&hl=fr#c667bae7bdc59473

The only way I’ve
found to solve this problem is to make asterisk running under the user you want
to access to the messages. You have to change add some permissions to Asterisk
folders, and can use ACL for that in this way : 

 

setfacl -m u:wwwrun:rwX
/var/run 
setfacl -m u:wwwrun:rwX /usr/sbin/asterisk 
setfacl -Rm u:wwwrun:rwX /var/log/asterisk/ 
setfacl -Rm u:wwwrun:rwX /var/log/asterisk/ 
setfacl -Rm u:wwwrun:rwX /var/spool/asterisk/ 
setfacl -Rm u:wwwrun:rwX /var/lib/asterisk/

 

Then in this exemple you
can run asterisk with wwwrun user.

 

An other way is to launch
a script Perl for example in the dialplan which check files/folsers permission when
appends a hang up to any channel.

 

Hope that can help you

 

Salim

 









De :
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Scott Keagy
Envoyé : mardi, 7. novembre
2006 20:22
À : Asterisk Users Mailing
List - Non-Commercial Discussion
Objet : [asterisk-users]
loosening voicemail file permissions formsg.txt and msg.wav



 

HI folks,

 

I figured out where in the source code to hack the
.wav file permissions which were set too restrictive for me, but I can’t
figure out how to do the same for the .txt file.

 

Looks like the voicemail.c file sets it nicely for
asterisk1.4beta3 using a #define statement early on, but msg.txt
 comes out with permissions 0600 and there are no umask entries that
affect how asterisk is started (if anything it would be 022).

 

I’ve grepped through the entire source tree
from the expanded tarball, and changed every place where it says 0600 to 0666
and recompiled, but still no luck. I’ve grepped for umask entries like
077 that might cause the problem, but again no luck.

 

My work-around right now is a cron-job that chmods
these directories every minute, but this is ridiculous.

 

Anyone solved this?

 

Thanks,

Scott

 






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Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

[RR]

On 11/8/06, Steve Edwards <[EMAIL PROTECTED]> wrote:

I have alaw, g729, gsm, ulaw, and wav sound file sets so that should
cover the transcoding bases pretty well.



It should but if you're not allowing anything but ulaw all of those
are probably not ever being used. What you might want to have is slin
encoded prompts, which you didn't mention if you have or not. Although
not sure if even that will help in cases where, say If you allow
people to record their names which are then played upon entry and exit
of a participant of the conference, I would think they're recorded in
slin format (it being the default format for (*)), there is your
codec-xlate right there. I wonder if there's some place where we can
control what codec format is used to record these names and stuff in
meetme. If this can be controlled and we can standardize the entire
thing with ulaw, that should help a LOT, I'd think. And
creation/termination of conferences could also be causing CPU usage,
all adding upto the spikes.

Butwhat do I know, maybe a digium person can shed more light on it.
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Re: [asterisk-users] Follow Me problems

[Philippe Lindheimer]

<[EMAIL PROTECTED]>From: "Time Bandit" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion" Date: Tue, 7 Nov 2006 08:53:51 -0500Subject: Re: [asterisk-users] Follow Me problems > Today we appear to have discovered our first bug.  We have an extension> setup to "followme" by ringing that extension + an external cell #> (ringall).  If nobody answers after 20 seconds the "destination if no> answer" is set to go to the extensions voicemail in the "followme" module.> The problem is it just keeps ringing forever.  If we delete the followme it> forwards to the voicemail as per the default SIP extension configuration> with voicemail enabled.>> Anyone run into this?  Is there a workaround?  Any advice would be
 greatly> appreciated as always.>> Our configuration is:> Supermicro Pentium D 2.66 Server with 2x512MB Memory> 3ware 8006-2LP Hardware RAID 1> Sangoma A200D with 8fxo (latest firmware/drivers as of last week)> CentOS 4.4> Asterisk 1.2.13> Zaptel 1.2.10> FreePBX 2.1.3When Asterisk dial the Cell phone, it goes out on the ZAP channel(Sangoma A200D), so as soon as it hit that channel, the call isconsidered answered even if the cell phone never actually pickup thecall. I didn't play with the "followme" module myself but that is whatI suspect is happening. Just watch the console and you should seesomething like "Zap/1-1 answered ..."That sound probably correct. Take a look at 2.2 beta2. You can choose to require confirmation on external numbers (press 1 to accept, 2 to decline). That way it is not treated as answered. It works very very well
 and has some additional enhancements.philippe<[EMAIL PROTECTED]><[EMAIL PROTECTED]> 


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[asterisk-users] RxFAX - How to catch errors in the dialplan

[Håkan Källberg]

Hello!

We use RxFAX, version 0.0.2-pre26 over ISDN and it mostly works
well. Occasionally we get .tiff files that are not convertible
with tiff2pdf. This may occur if the connection is interrupted,
I guess, but we also saw cases where the sender seems to
have got an OK but the .tiff is not readable anyway. Therefor I'd
like to check the return of the rxfax command a bit closer,
but there is no return value variable. The application is
said to return 0 or -1. The -1 should mean that the "user"
hanged up. Which user, sender or receiver? What was the
reason of the hangup? Error or success? And what about 0? Error
or success? I am confused!

Does anyone have a working dialplan to catch error/success
from RxFAX? If an error occurred I don't have to worry about
corrupted .tiffs.  A return STATUS variable would be most
welcome!

I saw one dialplan example with RxFAX and priority jumping,
where on jump to n+100 a retry, goto(n) was made. Does that
make sense? I have seen no indication that RxFAX should support
priority jumping or a j option. We use * 1.2.13.

What about the 0.0.3-preX series? It was always said to be
experimental and the 0.0.2 more for production use. Is this
still valid, or should I try the newer 0.0.3 versions? We use
current mISDN/chan_misdn, * 1.2.13, Linux kernel 2.6.13.

Mostly RxFAX wokrs very well, and the users are happy with it!

Best Regards: Håkan



pgprpTGFWJLw3.pgp
Description: PGP signature
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Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

[Steve Edwards]

On Wed, 8 Nov 2006, RR wrote:


On 11/8/06, Steve Edwards <[EMAIL PROTECTED]> wrote:


"top" refreshing every 3 seconds shows the asterisk process consuming from
10% to 70% of the CPU. "top" refreshing every 30 seconds shows around 30%.
Does anybody know what causes the "spikes."


Yeah I'd be interested to know as well. I wonder if creation/tear-down
of sessions does that. A conference in session should eventually get
to a stable CPU consumption. You might want to have a test system and
either through sipsak or manually create a bunch of conferences and
watch the CPU. If you're playing the entry/exit sounds, recoding and
announcing names, playing participant counts and all of these are
non-ulaw encoded prompts etc. you will get those spikes as that's
where codec-translation will happen.


I have alaw, g729, gsm, ulaw, and wav sound file sets so that should 
cover the transcoding bases pretty well.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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RE: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

[Steve Edwards]

I think you're getting confused with the top/bottom posting or something. 
I'll reformat in the order the posts were posted.



On 11/2/06, Steve Edwards <[EMAIL PROTECTED]> wrote:

I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application
is mostly meetme conferences being created and closed all day long.
Peak load is around 200 SIP calls.

I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I
haven't had a crash since. Meetme does not play well with SMP.



From: RR [mailto:[EMAIL PROTECTED]

I have constantly got conflicting reports about meetme and can't really make
up my mind to actually put meetme into service till I find something better
or just stick with meetme and be happy? I like the features it has but
performance wise I have heard all sorts of things, yours being the most
positive so far. so just wondering if I can learn something from you.


On Tue, 7 Nov 2006, shadowym wrote:


Steve,

I am running EXACTLY the same software as you.  SMP kernel and all.  Are you
using the IDENTICAL non-SMP kernel verision# ?  No crashes at all here for
the 2 weeks it has been up but we don't use meetme.


I complained about constant crashes in meetme that disappeared with a 
simple reboot to a non-SMP kernel (yes, the exact same version number).


RR replied with questions about my meetme configuration.

You (shadowym) replied saying you're running an SMP kernel, but not 
meetme.


I'm not surprised you're not crashing. It's the combination of SMP and 
meetme that is unstable. Others have said they are running SMP and meetme 
and not crashing so maybe the HP DL380's have some unique problem with 
SMP. Or maybe their callers are not constantly creating and closing 
conferences. Most of the crashes appeared to be when a channel or a 
conference was free'd.


I tried with every version from 1.2.9 through 1.2.13.1 and several SVN's 
in between. Not a single configuration was stable until the light bulb 
when on and I tried a non-SMP kernel. Not a single crash since.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

[RR]

On 11/8/06, Steve Edwards <[EMAIL PROTECTED]> wrote:

All calls come in from a Tekelec 7000 via SIP.

Out of a peak of 200 calls, probably around 100 are in meetme, others are
listening to recorded messages or bouncing around in the menus.


Sounds exactly like what people in my system would be doing.



No OS tweaks, no Asterisk source tweaks.

A TE410p is used as a timing source. The sound quality was not acceptable
with ztdummy.


Aha, so that's something I don't have and most prob. can't have (no
empty PCI slots left on the 1U servers). Hmmm maybe that might make
the difference between how many conferences my boxes will handle
before it starts to sound bad!


I stripped down /etc/asterisk/modules.conf just 'cause "parts left out
don't get broken" :)


Agreed, I have even removed non-used conf files so the size of (*) in
memory is significantly smaller.


My sip.conf only allows ulaw, but "show channel" shows some using ulaw and
some using slin. This may be changing as the calls bounce from meetme to
recorded wav messages. The Zap pseudo channels show ulaw -- I would have
expected slin. Somebody who understands codec switching could help out and
explain it to both of us :)


Think you would only see slin if some system playback needs to access
non-ulaw encoded files or users come on a different codec than others.
Since the latter isn't happening, there's no need for your system to
convert anything to slin, which is why your systems shows the Zap
pseudo channels as ulaw and playback of recorded messages doesn't use
the Zap pseudo channels. So unless my understanding is wrong, what
your systems shows is consistent with your description of the settings
you have there :)


"top" refreshing every 3 seconds shows the asterisk process consuming from
10% to 70% of the CPU. "top" refreshing every 30 seconds shows around 30%.
Does anybody know what causes the "spikes."


Yeah I'd be interested to know as well. I wonder if creation/tear-down
of sessions does that. A conference in session should eventually get
to a stable CPU consumption. You might want to have a test system and
either through sipsak or manually create a bunch of conferences and
watch the CPU. If you're playing the entry/exit sounds, recoding and
announcing names, playing participant counts and all of these are
non-ulaw encoded prompts etc. you will get those spikes as that's
where codec-translation will happen.
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RE: [asterisk-users] "Sticky" Polycom 501 keys and handset

[Darryl Dunkin]

This should 
be in your Asterisk sip_notify.conf file by default I believe (if not, add it 
with an appropriate name):
[polycom-check-cfg]Event=>check-syncContent-Length=>0
 
Then in the 
Asterisk run this (assuming the phone is registered 
properly):
sip notify 
polycom-check-cfg 
 
If the 
configuration on your FTP server (assuming FTP/TFTP configuration) has changed, 
it will reboot.
 
Otherwise, in 
your sip.cfg for your phones, look for this:
voIpProt.SIP.specialEvent.checkSync.alwaysReboot="0"
 
Change it to this to always 
reboot when receiving the notify:
voIpProt.SIP.specialEvent.checkSync.alwaysReboot="1"
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Rick 
SmithSent: Tuesday, November 07, 2006 17:44To: 'Asterisk 
Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[asterisk-users] "Sticky" Polycom 501 keys and handset


hmm, 
I’d like to know that.  How do you reboot remotely ? J
 


From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
MikeSent: Tuesday, November 07, 2006 2:13 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[asterisk-users] "Sticky" Polycom 501 keys and 
handset
 
Disregard 
my previous message, I succeeded in downgrading my phones.  And it worked, 
thanks Rick for the info.  Is there any Polycom-specific mailing list I 
should be on to be aware of stuff like that?
 
Also, 
would you know how to check the version of sip.ld remotely? I know how to reboot 
remotely, and I did for a few phones, but my paranoid self would like to double 
check and see if the sip.ld 1.6.7 re-installed ok by checking the current 
version.  Is that even possible?
 
Mike

   
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Rick 
  SmithSent: November 7, 2006 11:28 AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE: 
  [asterisk-users] "Sticky" Polycom 501 keys and handset
  I 
  had this EXACT same problem, and 2.0.x is the problem according to Polycom 
  Tech Support.
   
  I 
  had such a hard time explaining the problem, too…
   
  Downgraded 
  to 1.6.7 and all worked well again.  Polycom says if you’re using 
  Asterisk, don’t
  go 
  past 1.6.7 until they say to.
   
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  MikeSent: Tuesday, November 07, 2006 11:02 AMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  [asterisk-users] "Sticky" Polycom 501 keys and 
  handset
   
  
  Hi,
  
   
  
  I've recently 
  bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 
  2.0.1.   I just noticed something, which I first blamed on Asterisk 
  and NATs (a 2 second silence at the beginning of a call).  Something 
  I've noticed also on my old phone (which is having the same problem now, 
  but its also been upgraded).
  
   
  
  My keys are 
  sticky.  Simple as that.  Sometimes I press a number and the key 
  comes up (the hardware seems fine) but the phone produces this lng tone as 
  if I had pressed the key for 3 seconds.  Even the receiver is sticky, 
  giving my dialtone when I lift it only 1-2 seconds after I lift the 
  handset.  It simply looks like the phone can't keep up, like a 
  sluggish computer.
  
   
  
  Anybody has ever 
  seem this?  I'd like to downgrde to SIP 1.6.7 to see if the new sip app 
  was the problem.  How can I do that?  I've placed the old sip.ld 
  file where I had to, but the phone wont pick it up.  
  
  
   
  
  Short of that, can 
  somebody point me to the newest firmware (2.0.2) to see if that would 
  help?
  
   
  
  Mike 
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Re: [asterisk-users] Re: Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

[Enrique Martinez]

AFAIK, in 1.2.x the insecure=very change in favor to insecure=port,invite,

also you can try with allowguest=yes

Regards

JR Richardson wrote:

Update,

I loaded asterisk 1.0.10 and it worked straight away. I can send
unauthenticated calls to asterisk.  Something in 1.2.9.1 and 1.2.13
are not allowing unauthenticated calls when insecure=very is set in
sip.conf, either in the global or peer context.

Are there any switches in the Asterisk Makefile to allow this?

JR

On 11/7/06, JR Richardson <[EMAIL PROTECTED]> wrote:

Hi All,

I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:

asterisk sip >< sip TNT pri >< pri asterisk

The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1.  I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the SIP Asterisk server. I
have tried many variations of using sip options insecure,
autocreatepeer, permit/deny, host, user, etc but can't seem to get
asterisk to accept an unauthenticated call from the TNT using SIP.  I
keep getting SIP/2.0 407 Proxy Authentication Required.  I know others
have done this, but with older Asterisk versions, I'm wondering what
versions of Asterisk are known to work with the MAX TNT and with what
version of the TNT?

I'm confident this is an asterisk issue, with insecure=very, I should
be able to pass calls to asterisk without trying to authenticate it
first.  But this is not happening.

Here is a debug of a call and a snip from my sip.conf:

sip.conf

[maxtnt]
type=friend
host=10.10.14.131
insecure=very
dtmfmode=inband
callerid="MaxTNT" 
context=trunktntin
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw

debug

lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
Remote-Party-Id: "NO CID NAME"
;screen=no;id-type=subscriber;party=calling;privacy=off 


i: [EMAIL PROTECTED]
CSeq: 639089 INVITE
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
m: 
k: replaces
c: application/sdp
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
l: 232

v=0
o=t1gw01 531756636 531756636 IN IP4 10.10.14.131
s=Session SDP
c=IN IP4 10.10.14.131
t=0 0
m=audio 40198 RTP/AVP 0 96
a=silenceSupp:on
a=ecan:b on g168
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=rtpmap:0 PCMU/8000

--- (16 headers 11 lines) ---
Using INVITE request as basis request - 
[EMAIL PROTECTED]

Sending to 10.10.14.131 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.10.14.131:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131
From: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
To: ;tag=as41f8454e
Call-ID: [EMAIL PROTECTED]
CSeq: 639089 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
nonce="3ea7e98a"

Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 
15000 ms

Found user '1239'
lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   ;tag=as41f8454e
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
i: [EMAIL PROTECTED]
CSeq: 639089 ACK
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
User-Agent: Lucent-Universal-Gateway
l: 0


Any guidance will be much appreciated.

Thanks.

JR

--
JR Richardson
Engineering for the Masses






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Re: [asterisk-users] hicecaller ID

[William Piper]

To hide the caller ID, do this:
 
exten => _9NXXNXX.,1,Set(CALLERID(all)=Unknown<00>)
exten => _9NXXNXX.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED] 
bp 
On 11/7/06, Nik Engel <[EMAIL PROTECTED]> wrote:
Hi all !I have a question regarding flexible callerid settingusing the misdnI want to acheive the following:
when starting a call with 0 I want to display CALLERID (which is setupright now) but when I start the call with 9 I want the callerid to besurpressed.How can this be done?nik
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Re: [asterisk-users] Why dont my messages get through

[Andrew Furey]

On 11/8/06, Nick Hoffman <[EMAIL PROTECTED]> wrote:

> They do get through. Messages you send to the list won't get sent back
> to you, because you sent them.

Hi Alex. Are you sure about that? I receive a copy of every email I send to
the list.


I think it's just Gmail that hides them, especially since you have the
one you sent already there. It does the same for another mailing list
I'm on. This doesn't apply to Christian, of course.

(Interesting experiment would be to delete your sent copy as soon as
you send it, before the list server sends it back, and see if it
reappears...)

Andrew

--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
 -- Bill Garrett
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Re: [asterisk-users] Why dont my messages get through

[Dovid B]

I have seen this mainly with gmail. the logic is 
why do you need your own postings. Fish around to see if there is a setting in 
Gmail where it will keep the email. I know for myself I want the email's that I 
sent. It lets me know that they went out as well as it helps for sorting the 
emails.
 

  - Original Message - 
  From: 
  Alex 
  Robar 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, November 08, 2006 5:08 
  AM
  Subject: Re: [asterisk-users] Why dont my 
  messages get through
  They do get through. Messages you send to the list won't get 
  sent back to you, because you sent them. 
  On 11/7/06, Christian < [EMAIL PROTECTED]> 
  wrote:
  Hi,My 
messages to the list don't get through. This must be the tenth message i am 
trying to send! Please ignore this test 
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visit:   http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] 
  
  

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Re: [asterisk-users] Snom 360 flickering screen

[Nick Hoffman]

> - Original Message -
> From: "Nick Hoffman" <[EMAIL PROTECTED]>
> To: "asterisk-users Mailing List" 
> Sent: Tuesday, November 07, 2006 10:53 AM
> Subject: [asterisk-users] Snom 360 flickering screen
>
> > Hi guys. I just bought and configured a Snom 360 and have noticed that
> > the LCD is constantly flickering at a rate of ~10-15Hz (that's a
> > guess). Either way, it's very distracting. Has anyone else encountered
> > this before? Any solutions?
> >
> > Cheers,
> > -- Nick
> > E: [EMAIL PROTECTED]
> > P: +61 7 5591 3588
> > F: +61 7 5591 6588
> >
> > If you receive this email by mistake, please notify us and do not make
> > any use of the email.  We do not waive any privilege, confidentiality
> > or copyright associated with it.


On Wed November 8 2006 01:30, "Dovid B" <[EMAIL PROTECTED]> wrote:
> I had a problem with snom where the screen went completly blank. Snom
> told me there was an issue where that the cable going from the phone
> board to the screen would fall out. I opend the phone and sliped it back
> in.


Hi Dovid, thanks for the recommendation. I opened up my 360 and looked 
around. Everything was connected properly, but I noticed that some of the 
wires connecting the two PCBs were partially crushed by one of the case's 
support posts. When closing the case, I made sure to move the wires out of 
the way of the posts. The screen's much better now. If I look at it from 
an extreme angle I can see a lot of flickering, but at normal angles 
there's almost no flickering.

Thanks again!
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F: +61 7 5591 6588

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RE: [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer

[Curt Shaffer]

Take a look at OVA….

 

mms://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Outlook_Voice_Access_300k.wmv

 

 

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar
Sent: Tuesday, November 07, 2006
9:13 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Microsoft will enter VoIP market in earnest nextyear, says Ballmer



 

Unified messaging would
be nice. Not just having my VM's e-mailed to me, but to be able to manage them
from with Outlook (or any other mail client for that matter) would be nice. I
picture it sort of like an IMAP mailbox, and the mail client just has some kind
of functionality to recognize that the message is a VM and not a mail message
(so it could display length, date/time received, CID, and provide a
"play" button). 

Just my two cents.

Alex



On 11/7/06, Dean
Collins <[EMAIL PROTECTED]>
wrote:





http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm

 

There's
not much in the article so only click through if super interested but I'm
curious and looking for people's opinions.

 

What
application integration would you like to see between MS (either Office or
other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook
and number pop I'm kind of curious what other functionality there is to be
developed (I'd also like to see drop and drag from outlook into conference
calls.

 

 

 

What
would you like to see in asterisk, if we get some solid responses we'll see
about organizing some bounties to get it developed.

 

 

 



Regards, 

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial). 

 



 






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-- 
Alex Robar
[EMAIL PROTECTED] 






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Re: [asterisk-users] Fax & Digium

[Dovid B]

- Original Message - 
From: "Hadley Rich" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, November 08, 2006 5:07 AM
Subject: Re: [asterisk-users] Fax & Digium


On Wednesday 08 November 2006 13:15, Ken Williams wrote:

I was planning on using a TDM400P with 3 FXO & 1 FXS, with the 1 FXS
being used for a fax machine. It now appears that Digium doesn't
support this, are there other manufacturers anyone can recommend that
will support it? Has anyone used a TDM400P in this setup and had it
work without much issue?


Yes, I have implemented this on a few occasions and it has worked fine for 
me.


hads

--
http://nicegear.co.nz
New Zealand's VoIP supplier
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I tried this and no I have a big percentage of faxes that do not go thru. Do 
you mind sharing how you set it up ?


Thanks. 



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Re: [asterisk-users] Help with latest Asterisk on latest Debian

[Tzafrir Cohen]

On Wed, Nov 08, 2006 at 02:24:23AM +0100, Christian wrote:
> Hi all,
> I am running latest test version of Debian and I have done the following:
> First, i did apt-get build-dep asterisk
> Then I downloaded the latest version of Zaptel, Libpri and Asterisk, 
> something with 1.4.
> I am using Kernel 2.6.17.686-2 so I did

from package, or self-built?

> apt-get install linux-headers-2.6.17-2-686

Was it properly installed, with all dependencies?

> However, I get the following error message when trying to compile Zaptel. The 
> scripts folder seem to be missing in /usr/src/linux-headers-2.6.17-2-686
> It is there, but to me it looks like a file not a folder.
> make -C /lib/modules/2.6.17-2-686/build SUBDIRS=/root/zaptel-1.4.0-beta2 
> modules
> make[1]: Entering directory `/usr/src/linux-headers-2.6.17-2-686'
> Makefile:266: /usr/src/linux-headers-2.6.17-2-686/scripts/Kbuild.include: No 
> such file or directory
> /bin/sh: line 0: [: -lt: unary operator expected
> make[1]: *** No rule to make target 
> `/usr/src/linux-headers-2.6.17-2-686/scripts/Kbuild.include'.  Stop.
> make[1]: Leaving directory `/usr/src/linux-headers-2.6.17-2-686'
> make: *** [linux26] Error 2
> I am a little new to this as well.
> All the best and many thanks for all your help,
> Christian

One other thing to try:

  apt-get install zaptel-source
  m-a -t -i build zaptel

This would be zaptel 1.2, but similar enough.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Why dont my messages get through

[Alex Robar]

I _was_ sure until mention it just now... I certainly don't get a copy of any messages I sent to the list, whether I send  from my personal or office accounts. Maybe the way my mail clients are handling it? If so, my apologies to Christian.
AlexOn 11/7/06, Nick Hoffman <[EMAIL PROTECTED]> wrote:
On 11/7/06, Christian <[EMAIL PROTECTED]> wrote:> > Hi,> > My messages to the list don't get through. This must be the tenth> > message i am trying to send!
> > Please ignore this test message.On Wed November 8 2006 13:08, "Alex Robar" <[EMAIL PROTECTED]> wrote:> They do get through. Messages you send to the list won't get sent back
> to you, because you sent them.Hi Alex. Are you sure about that? I receive a copy of every email I send tothe list.-- NickE: [EMAIL PROTECTED]
P: +61 7 5591 3588F: +61 7 5591 6588If you receive this email by mistake, please notify us and do not make anyuse of the email.  We do not waive any privilege, confidentiality orcopyright associated with it.
-- Alex Robar[EMAIL PROTECTED]
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Re: [asterisk-users] Why dont my messages get through

[Nick Hoffman]

On 11/7/06, Christian <[EMAIL PROTECTED]> wrote:
> > Hi,
> > My messages to the list don't get through. This must be the tenth
> > message i am trying to send!
> > Please ignore this test message.

On Wed November 8 2006 13:08, "Alex Robar" <[EMAIL PROTECTED]> wrote:
> They do get through. Messages you send to the list won't get sent back
> to you, because you sent them.

Hi Alex. Are you sure about that? I receive a copy of every email I send to 
the list.
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
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Re: [asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer

[Alex Robar]

Unified messaging would be nice. Not just having my VM's e-mailed to me, but to be able to manage them from with Outlook (or any other mail client for that matter) would be nice. I picture it sort of like an IMAP mailbox, and the mail client just has some kind of functionality to recognize that the message is a VM and not a mail message (so it could display length, date/time received, CID, and provide a "play" button).
Just my two cents.AlexOn 11/7/06, Dean Collins <[EMAIL PROTECTED]> wrote:

















http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm

 

There's not much in the article so only click through
if super interested but I'm curious and looking for people's
opinions.

 

What application integration would you like to see between
MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from
dial from outlook and number pop I'm kind of curious what other
functionality there is to be developed (I'd also like to see drop and
drag from outlook into conference calls.

 

 

 

What would you like to see in asterisk, if we get some solid
responses we'll see about organizing some bounties to get it developed.

 

 

 



Regards,


Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]

+1-212-203-4357
Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).


 



 







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Re: [asterisk-users] Why dont my messages get through

[Alex Robar]

They do get through. Messages you send to the list won't get sent back to you, because you sent them. On 11/7/06, Christian <
[EMAIL PROTECTED]> wrote:Hi,My messages to the list don't get through. This must be the tenth message i am trying to send!
Please ignore this test message.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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Alex Robar[EMAIL PROTECTED]
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Re: [asterisk-users] Fax & Digium

[Hadley Rich]

On Wednesday 08 November 2006 13:15, Ken Williams wrote:
> I was planning on using a TDM400P with 3 FXO & 1 FXS, with the 1 FXS
> being used for a fax machine.  It now appears that Digium doesn't
> support this, are there other manufacturers anyone can recommend that
> will support it?  Has anyone used a TDM400P in this setup and had it
> work without much issue?

Yes, I have implemented this on a few occasions and it has worked fine for me.

hads

-- 
http://nicegear.co.nz
New Zealand's VoIP supplier
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Re: [asterisk-users] Dial plan Question

[Doug Crompton]

Answering my own question. If you want to connect an spa3K with
generic pstn inbound do the following...

for the pstn to voip dialplan in the pstn tab -  (S0<:ip-address-of-*>)

in sip.conf

[sipurafxo]
context=from-pstn
etc.

Then in * extensions.conf use the s extension.

[from-pstn]
exten => s,1,answer()
exten => s,2,dial.
etc.

Makes it alot easier as you do not have to deal with extension matching
when you know where it is coming from.

Doug


On Tue, 7 Nov 2006, Doug Crompton wrote:

> Thanks, that set off a light bulb In my spa3K my incoming dialplan was
> set to  (S0<:405>)
>
> Since this is a one FXO unit and my [from-pstn] will always be that line
> can I make it generic and use the 's' extension as I described? If so what
> would that spa3k dialplan be? just s0 ?
>
> Doug
>
> On Tue, 7 Nov 2006, Anselm Martin Hoffmeister wrote:
>
> > Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
> > > I am trying to do something that I see describe in a book and it is not
> > > working
> > >
> > > In my sip.conf, I have in my [fxo] context=from-pstn
> > >
> > > I then have in extensions.conf
> > >
> > > [from-pstn]
> > >
> > > exten s,1,answer()
> > > exten s,2,playback(blah)
> > >
> > > etc.
> > >
> > > It never answers but if I do this
> > >
> > > [from-pstn]
> > >
> > > exten _x.,1,answer()
> > > exten _x.,2,playback(blah)
> > >
> > > it works.  Why does the 's' extension not work here?
> >
> > If "fxo" means your SIP provider, and you register with him, a specific
> > extension will be called. Which one shall be called can be selected by
> > the last parameter of the register statement, e.g.
> >
> > register => 075741:[EMAIL PROTECTED]:5060/492281234567
> >
> > will cause the incoming calls to appear in extension 492281234567.
> >
> > Comes in handy if you have several accounts with a single SIP provider:
> > This way, you can simply distinguish the outward phone number for which
> > the call came in.
> >
> > BR
> > Anselm
> >
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> "Those that sacrifice essential liberty to obtain a little temporary safety
>  deserve neither liberty nor safety."  -- Ben Franklin (1759)
>
> 
> *  Doug Crompton *
> *  Richboro, PA 18954*
> *  215-431-6307  *
> **
> * [EMAIL PROTECTED]*
> * http://www.crompton.com  *
> 
>
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>


"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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RE: [asterisk-users] "Sticky" Polycom 501 keys and handset

[Darryl Dunkin]

In Asterisk enter 'sip show peer ' and you can see this in 
the Useragent field.
 
Example (for 2.0.1):
Useragent    : 
PolycomSoundPointIP-SPIP_501-UA/2.0.1.0313


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
MikeSent: Tuesday, November 07, 2006 11:13To: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[asterisk-users] "Sticky" Polycom 501 keys and handset

Disregard my previous message, I succeeded in downgrading 
my phones.  And it worked, thanks Rick for the info.  Is there any 
Polycom-specific mailing list I should be on to be aware of stuff like 
that?
 
Also, would you know how to check the version of sip.ld 
remotely? I know how to reboot remotely, and I did for a few phones, but my 
paranoid self would like to double check and see if the sip.ld 1.6.7 
re-installed ok by checking the current version.  Is that even 
possible?
 
Mike

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Rick 
  SmithSent: November 7, 2006 11:28 AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE: 
  [asterisk-users] "Sticky" Polycom 501 keys and handset
  
  
  I 
  had this EXACT same problem, and 2.0.x is the problem according to Polycom 
  Tech Support.
   
  I 
  had such a hard time explaining the problem, too…
   
  Downgraded 
  to 1.6.7 and all worked well again.  Polycom says if you’re using 
  Asterisk, don’t
  go 
  past 1.6.7 until they say to.
   
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  MikeSent: Tuesday, November 07, 2006 11:02 AMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  [asterisk-users] "Sticky" Polycom 501 keys and 
  handset
   
  
  Hi,
  
   
  
  I've recently 
  bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 
  2.0.1.   I just noticed something, which I first blamed on Asterisk 
  and NATs (a 2 second silence at the beginning of a call).  Something 
  I've noticed also on my old phone (which is having the same problem now, 
  but its also been upgraded).
  
   
  
  My keys are 
  sticky.  Simple as that.  Sometimes I press a number and the key 
  comes up (the hardware seems fine) but the phone produces this lng tone as 
  if I had pressed the key for 3 seconds.  Even the receiver is sticky, 
  giving my dialtone when I lift it only 1-2 seconds after I lift the 
  handset.  It simply looks like the phone can't keep up, like a 
  sluggish computer.
  
   
  
  Anybody has ever 
  seem this?  I'd like to downgrde to SIP 1.6.7 to see if the new sip app 
  was the problem.  How can I do that?  I've placed the old sip.ld 
  file where I had to, but the phone wont pick it up.  
  
  
   
  
  Short of that, can 
  somebody point me to the newest firmware (2.0.2) to see if that would 
  help?
  
   
  
  Mike 
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Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

[Scott Keagy]

Title: Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue






When all else fails I resort to adding this in the sip.conf peer config:

Insecure=invite,port

It took me a while to figure out they can be used together.

Regards,
Scott

- Original Message -
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com 
Sent: Tue Nov 07 15:23:26 2006
Subject: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue

Hi All,

I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:

asterisk sip >< sip TNT pri >< pri asterisk

The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1.  I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the SIP Asterisk server. I
have tried many variations of using sip options insecure,
autocreatepeer, permit/deny, host, user, etc but can't seem to get
asterisk to accept an unauthenticated call from the TNT using SIP.  I
keep getting SIP/2.0 407 Proxy Authentication Required.  I know others
have done this, but with older Asterisk versions, I'm wondering what
versions of Asterisk are known to work with the MAX TNT and with what
version of the TNT?

I'm confident this is an asterisk issue, with insecure=very, I should
be able to pass calls to asterisk without trying to authenticate it
first.  But this is not happening.

Here is a debug of a call and a snip from my sip.conf:

sip.conf

[maxtnt]
type=friend
host=10.10.14.131
insecure=very
dtmfmode=inband
callerid="MaxTNT" 
context=trunktntin
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw

debug

lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
Remote-Party-Id: "NO CID NAME"
;screen=no;id-type=subscriber;party=calling;privacy=off
i: [EMAIL PROTECTED]
CSeq: 639089 INVITE
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
m: 
k: replaces
c: application/sdp
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
l: 232

v=0
o=t1gw01 531756636 531756636 IN IP4 10.10.14.131
s=Session SDP
c=IN IP4 10.10.14.131
t=0 0
m=audio 40198 RTP/AVP 0 96
a=silenceSupp:on
a=ecan:b on g168
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=rtpmap:0 PCMU/8000

--- (16 headers 11 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 10.10.14.131 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.10.14.131:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131
From: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
To: ;tag=as41f8454e
Call-ID: [EMAIL PROTECTED]
CSeq: 639089 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a"
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Found user '1239'
lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   ;tag=as41f8454e
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
i: [EMAIL PROTECTED]
CSeq: 639089 ACK
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
User-Agent: Lucent-Universal-Gateway
l: 0


Any guidance will be much appreciated.

Thanks.

JR

--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

[Barry Fawthrop]

what is the sip.conf for   1239
which I'm going to assume is a extension on the TNT

Barry

JR Richardson wrote:

Hi All,

I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:

asterisk sip >< sip TNT pri >< pri asterisk

The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1.  I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the SIP Asterisk server. I
have tried many variations of using sip options insecure,
autocreatepeer, permit/deny, host, user, etc but can't seem to get
asterisk to accept an unauthenticated call from the TNT using SIP.  I
keep getting SIP/2.0 407 Proxy Authentication Required.  I know others
have done this, but with older Asterisk versions, I'm wondering what
versions of Asterisk are known to work with the MAX TNT and with what
version of the TNT?

I'm confident this is an asterisk issue, with insecure=very, I should
be able to pass calls to asterisk without trying to authenticate it
first.  But this is not happening.

Here is a debug of a call and a snip from my sip.conf:

sip.conf

[maxtnt]
type=friend
host=10.10.14.131
insecure=very
dtmfmode=inband
callerid="MaxTNT" 
context=trunktntin
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw

debug

lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
Remote-Party-Id: "NO CID NAME"
;screen=no;id-type=subscriber;party=calling;privacy=off 


i: [EMAIL PROTECTED]
CSeq: 639089 INVITE
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
m: 
k: replaces
c: application/sdp
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
l: 232

v=0
o=t1gw01 531756636 531756636 IN IP4 10.10.14.131
s=Session SDP
c=IN IP4 10.10.14.131
t=0 0
m=audio 40198 RTP/AVP 0 96
a=silenceSupp:on
a=ecan:b on g168
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=rtpmap:0 PCMU/8000

--- (16 headers 11 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 10.10.14.131 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.10.14.131:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131
From: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
To: ;tag=as41f8454e
Call-ID: [EMAIL PROTECTED]
CSeq: 639089 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
nonce="3ea7e98a"

Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 
15000 ms

Found user '1239'
lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   ;tag=as41f8454e
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
i: [EMAIL PROTECTED]
CSeq: 639089 ACK
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
User-Agent: Lucent-Universal-Gateway
l: 0


Any guidance will be much appreciated.

Thanks.

JR


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RE: [asterisk-users] "Sticky" Polycom 501 keys and handset

[Rick Smith]

hmm, I’d like to know that.  How do you reboot remotely ? J

 





From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, November 07, 2006 2:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and
handset





 

Disregard my previous message, I
succeeded in downgrading my phones.  And it worked, thanks Rick for the
info.  Is there any Polycom-specific mailing list I should be on to be
aware of stuff like that?

 

Also, would you know how to check
the version of sip.ld remotely? I know how to reboot remotely, and I did for a
few phones, but my paranoid self would like to double check and see if the
sip.ld 1.6.7 re-installed ok by checking the current version.  Is that
even possible?

 

Mike



 







From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Rick Smith
Sent: November 7, 2006 11:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] "Sticky" Polycom 501 keys and
handset

I had this EXACT same
problem, and 2.0.x is the problem according to Polycom Tech Support.

 

I had such a hard time
explaining the problem, too…

 

Downgraded to 1.6.7 and all
worked well again.  Polycom says if you’re using Asterisk, don’t

go past 1.6.7 until they say
to.

 





From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, November 07, 2006 11:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] "Sticky" Polycom 501 keys and
handset





 



Hi,





 





I've recently bought new Polycom 501 phones,
upgraded to bootrom 3.2.2 and SIP 2.0.1.   I just noticed something,
which I first blamed on Asterisk and NATs (a 2 second silence at the beginning
of a call).  Something I've noticed also on my old phone (which is
having the same problem now, but its also been upgraded).





 





My keys are sticky.  Simple as
that.  Sometimes I press a number and the key comes up (the hardware seems
fine) but the phone produces this lng tone as if I had pressed the key for
3 seconds.  Even the receiver is sticky, giving my dialtone when I lift it
only 1-2 seconds after I lift the handset.  It simply looks like the
phone can't keep up, like a sluggish computer.





 





Anybody has ever seem this?  I'd like to
downgrde to SIP 1.6.7 to see if the new sip app was the problem.  How can
I do that?  I've placed the old sip.ld file where I had to, but the phone
wont pick it up.  





 





Short of that, can somebody point me to the
newest firmware (2.0.2) to see if that would help?





 





Mike 










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[asterisk-users] test please ignore

[Christian]

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[asterisk-users] Help with latest Asterisk on latest Debian

[Christian]

Hi all,
First, I really hope that this message gets through since i had to change email 
address on this list. Only one message from me with my previous address got 
through.
I am running latest test version of Debian and I have done the following:
First, i did apt-get build-dep asterisk
Then I downloaded the latest version of Zaptel, Libpri and Asterisk, something 
with 1.4.
I am using Kernel 2.6.17.686-2 so I did
apt-get install linux-headers-2.6.17-2-686
However, I get the following error message when trying to compile Zaptel. The 
scripts folder seem to be missing in /usr/src/linux-headers-2.6.17-2-686
It is there, but to me it looks like a file not a folder.
make -C /lib/modules/2.6.17-2-686/build SUBDIRS=/root/zaptel-1.4.0-beta2 modules
make[1]: Entering directory `/usr/src/linux-headers-2.6.17-2-686'
Makefile:266: /usr/src/linux-headers-2.6.17-2-686/scripts/Kbuild.include: No 
such file or directory
/bin/sh: line 0: [: -lt: unary operator expected
make[1]: *** No rule to make target 
`/usr/src/linux-headers-2.6.17-2-686/scripts/Kbuild.include'.  Stop.
make[1]: Leaving directory `/usr/src/linux-headers-2.6.17-2-686'
make: *** [linux26] Error 2
I am a little new to this as well.
All the best and many thanks for all your help,
Christian


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[asterisk-users] RE: Queues and multiple lines

[Shane O'Cain]

I think you could enable call waiting (*70) on those stations and they would
have the second line ring in. This is what I have done in the past. The
second call would continue to use the ring strategy configured in the queue.


You can also enable call waiting from the Asterisk command line by typing:
database put CW XXX ENABLED (where XXX is the extension) 

Not sure if this is what you're looking for, but hope it helps.

Regards,
Shane


--

Message: 9
Date: Tue, 07 Nov 2006 12:39:25 -0600
From: Michael Sampson <[EMAIL PROTECTED]>
Subject: [asterisk-users] Queues and multiple lines
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Say I have agents using a softphone like eyebeam that has 6 lines. They 
log in to the queue. Say there are 3 agents in my queue. 3 calls come in 
and all three agents are on a call. Now a fourth call comes in. Is it 
possible to have it setup so that the 4 call rings on line 2 of one of 
my agents, if they don't get it within the time limit it rings on line 2 
of another agent and so on. An agent can then put their current call on 
hold and go to the new call, say something like "thanks for calling 
please hold", then go back to their first call, finish it up and then go 
back to the second call. I hope that made sense. I'm sure there is a way 
to get it done, but how flexible is the current queue system in Asterisk 
with stuff like this?

-- 
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

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RE: [asterisk-users] Queues and multiple lines

[brandon kruz]

Using SIP:

Just create another user account
say the softphones user's name is bob:

create [bob] (bob's main line on his softphone)
create [bob1] (same configuration options, then you can do
all your other configurations for this user )

hope this helps

anyone is open to correcting me :]

my 2 cents
`KruZ~



From: Michael Sampson <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: Asterisk Users Mailing List - Non-Commercial 
Discussion

Subject: [asterisk-users] Queues and multiple lines
Date: Tue, 07 Nov 2006 12:39:25 -0600
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Say I have agents using a softphone like eyebeam that has 6 lines. They log 
in to the queue. Say there are 3 agents in my queue. 3 calls come in and 
all three agents are on a call. Now a fourth call comes in. Is it possible 
to have it setup so that the 4 call rings on line 2 of one of my agents, if 
they don't get it within the time limit it rings on line 2 of another agent 
and so on. An agent can then put their current call on hold and go to the 
new call, say something like "thanks for calling please hold", then go back 
to their first call, finish it up and then go back to the second call. I 
hope that made sense. I'm sure there is a way to get it done, but how 
flexible is the current queue system in Asterisk with stuff like this?


--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

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_
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Microsoft Office Live 
http://clk.atdmt.com/MRT/go/mcrssaub0050001411mrt/direct/01/


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[asterisk-users] Glitches in sound every time that Asterisk receives reINVITEs

[rcarvalho]

Hi all,

My Asterisk server is working fine, although every time that in the middle of
any call there is a reinvite, the user hears a glitch. Why is this happening?
How can I solve this problem?

Thanks in advance,
Ricardo Carvalho.




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[asterisk-users] Fax & Digium

[Ken Williams]

I was planning on 
using a TDM400P with 3 FXO & 1 FXS, with the 1 FXS being used for a fax 
machine.  It now appears that Digium doesn't support this, are there other 
manufacturers anyone can recommend that will support it?  Has anyone used a 
TDM400P in this setup and had it work without much issue?
 
Thanks for the 
help,
Ken
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[asterisk-users] Why dont my messages get through

[Christian]

Hi,
My messages to the list don't get through. This must be the tenth message i am 
trying to send!
Please ignore this test message.

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[asterisk-users] [resolved] asterisk 1,4 and google talk

[Mani Sridhar]

hi,
it turns out that the iksemel library (which i installed using an rpm) was 
returning 0 when the function iks_has_tls() was called. it should return 1 
otherwise res_jabber.o thinks gnuTLS is not installed. i confirmed this by 
running a test program i wrote, that calls iks_has_tls . it returned 0.


i downloaded iksemel source, compiled it and now the test program returned 
1.


now, "jabber show connected" shows the google talk account as connected, but 
i don't see this buddy online on my other google talk buddy list.


i added an extension in extensions.conf that calls Gtalk/buddy, and as soon 
as i call this extension, asterisk terminates due to a segmentation fault. 
it didn't seem like a core was dumped - i'm still looking for it.


thanks
sridhar

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[asterisk-users] test message please ignore

[Christian]

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[asterisk-users] Microsoft will enter VoIP market in earnest next year, says Ballmer

[Dean Collins]

http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm

 

There’s not much in the article so only click through
if super interested but I’m curious and looking for people’s
opinions.

 

What application integration would you like to see between
MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from
dial from outlook and number pop I’m kind of curious what other
functionality there is to be developed (I’d also like to see drop and
drag from outlook into conference calls.

 

 

 

What would you like to see in asterisk, if we get some solid
responses we’ll see about organizing some bounties to get it developed.

 

 

 



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

 



 






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[asterisk-users] Re: Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

[JR Richardson]

Update,

I loaded asterisk 1.0.10 and it worked straight away. I can send
unauthenticated calls to asterisk.  Something in 1.2.9.1 and 1.2.13
are not allowing unauthenticated calls when insecure=very is set in
sip.conf, either in the global or peer context.

Are there any switches in the Asterisk Makefile to allow this?

JR

On 11/7/06, JR Richardson <[EMAIL PROTECTED]> wrote:

Hi All,

I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:

asterisk sip >< sip TNT pri >< pri asterisk

The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1.  I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the SIP Asterisk server. I
have tried many variations of using sip options insecure,
autocreatepeer, permit/deny, host, user, etc but can't seem to get
asterisk to accept an unauthenticated call from the TNT using SIP.  I
keep getting SIP/2.0 407 Proxy Authentication Required.  I know others
have done this, but with older Asterisk versions, I'm wondering what
versions of Asterisk are known to work with the MAX TNT and with what
version of the TNT?

I'm confident this is an asterisk issue, with insecure=very, I should
be able to pass calls to asterisk without trying to authenticate it
first.  But this is not happening.

Here is a debug of a call and a snip from my sip.conf:

sip.conf

[maxtnt]
type=friend
host=10.10.14.131
insecure=very
dtmfmode=inband
callerid="MaxTNT" 
context=trunktntin
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw

debug

lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
Remote-Party-Id: "NO CID NAME"
;screen=no;id-type=subscriber;party=calling;privacy=off
i: [EMAIL PROTECTED]
CSeq: 639089 INVITE
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
m: 
k: replaces
c: application/sdp
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
l: 232

v=0
o=t1gw01 531756636 531756636 IN IP4 10.10.14.131
s=Session SDP
c=IN IP4 10.10.14.131
t=0 0
m=audio 40198 RTP/AVP 0 96
a=silenceSupp:on
a=ecan:b on g168
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=rtpmap:0 PCMU/8000

--- (16 headers 11 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 10.10.14.131 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.10.14.131:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131
From: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
To: ;tag=as41f8454e
Call-ID: [EMAIL PROTECTED]
CSeq: 639089 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a"
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Found user '1239'
lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   ;tag=as41f8454e
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
i: [EMAIL PROTECTED]
CSeq: 639089 ACK
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
User-Agent: Lucent-Universal-Gateway
l: 0


Any guidance will be much appreciated.

Thanks.

JR

--
JR Richardson
Engineering for the Masses




--
JR Richardson
Engineering for the Masses
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[asterisk-users] astertest

[Victor Toofic]

Hi all!!

I've made some changes to the applications that Astertest was using to
monitor the performance of the server. Now is also possible to track the
bandwidth usage of the server, this has nothing to do with the executable
(astertest.exe) itself but with the events that the Asterisk Manager
generates.

The method described in:

http://www.asteriskguru.com/tutorials/astertest.html

to perform the test is still valid.

In the next days I am gonna make available some scripts to originate the
calls and to make some graphs of the test, just like astertest does ;)

You can find the sources here:

http://toofic.no-ip.org/pub/src/app_securax.tar.gz

I've compiled them against Asterisk 1.2.12.1, but I think there should not
be problems with other versions.

I hope someone could find it useful.

--
Grettings,
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Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

[Steve Edwards]

On Tue, 7 Nov 2006, RR wrote:


On 11/2/06, Steve Edwards <[EMAIL PROTECTED]> wrote:

I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application is
mostly meetme conferences being created and closed all day long. Peak load
is around 200 SIP calls.

I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I
haven't had a crash since. Meetme does not play well with SMP.


HI Steve,

I have constantly got conflicting reports about meetme and can't
really make up my mind to actually put meetme into service till I find
something better or just stick with meetme and be happy? I like the
features it has but performance wise I have heard all sorts of things,
yours being the most positive so far. so just wondering if I can learn
something from you.

So, is there anything special you've done in terms of configs,
modules, OS tweaking/tuning and the like, in other words, anything
over and above simply installing OS and (*) with meetme for the system
mentioned above? Have you standardised codecs across the board to
minimise translation overhead? If so, then what codec are you using?
Are all your users on IP or some can come through the PSTN via DIDs
etc?


All calls come in from a Tekelec 7000 via SIP.

Out of a peak of 200 calls, probably around 100 are in meetme, others are
listening to recorded messages or bouncing around in the menus.

No OS tweaks, no Asterisk source tweaks.

A TE410p is used as a timing source. The sound quality was not acceptable
with ztdummy.

I stripped down /etc/asterisk/modules.conf just 'cause "parts left out
don't get broken" :)

My sip.conf only allows ulaw, but "show channel" shows some using ulaw and
some using slin. This may be changing as the calls bounce from meetme to
recorded wav messages. The Zap pseudo channels show ulaw -- I would have
expected slin. Somebody who understands codec switching could help out and
explain it to both of us :)

"top" refreshing every 3 seconds shows the asterisk process consuming from
10% to 70% of the CPU. "top" refreshing every 30 seconds shows around 30%.
Does anybody know what causes the "spikes."

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] incoming call destination: IVR not working

[Mark Bryant]

Hi, I am a newbie putting together my first Asterisk system
and having a problem with the IVR handling incoming calls. 

 

I installed the Asterisk Trixbox version 1.2.2 with a X100P
FXO PCI card.  I have a PSTN line connected to the card.  I set up
two extensions: 200 and 201.  I created a test IVR that says, “for sales,
press 1, for support, press 2”.  If you press 1 it should go to
extension 200, if you press 2, it should go to extension 201.  I setup two
software SIP phones: SJ and Xlite (one for one extension and one for the other). 
If I dial  from the softphones, everything works as expected.  I can
also dial out and between the phones.  The problem is that if someone
dials in, the IVR does not respond to what they press (1 or 2) and after
repeating the greeting three times, hangs up.  If I change the incoming
destination to go directly to one of the extensions instead of the IVR, it
works fine.  I am guessing it has something to do with the IVR recognizing
the DTMF but not sure what to do from here.

 

TIA for any help or suggestions.

 

-mark

 






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Re: [asterisk-users] How do I make this stop? (Bridging of IAX channels?)

[Matt]

Asterisk is still going to try to native bridge the two channels. Once
this occurs chan_iax2 is going to notice that you don't want a native
transfer to happen and not do it.


Ok should it be giving me any indication that it has NOT done a
native transfer?  Or does it just say 'attempting native bridge',
indicating it is trying and then silently fails?
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RE: [asterisk-users] Question on Aastra phones and Astrisk

[shadowym]

Running several Aastra 9133i and 480CT phones with v1.4 firmware CentOS 4.4,
Asterisk 1.2.13, Zaptel 1.2.10, Freepbx2.1.3.  Using all default settings

I have not seen that problem. I am not exactly sure we are creating those
exact same conditions but it sounds like standard extension use to multiple
incoming calls correct?  That is all we are doing plus some more complicated
outgoing stuff.

-Original Message-
From: Matt [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, November 07, 2006 5:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk

*bump*  Anyone?

On 11/6/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:
> I wanted to add what we have both seen on traffic captures.
>
> You see Caller 1's RTP stream. Call 2 comes in and you see the 
> creation of its RTP stream. After Call 2 is put on hold the RTP stream 
> from Caller 1 disappears without a trace never to return and this is 
> when the one way audio is happening.
>
> And I also wanted to add that I am running 1.4.0 firmware for this phone.
>
> Thanks again!
>
>
>
> -Original Message-
> From: Curt Shaffer [mailto:[EMAIL PROTECTED]
> Sent: Monday, November 06, 2006 6:58 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk
>
> I'm the friend mentioned here.
>
> I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination 
> from the PBX to my provider. My issue has a slight twist to it but the 
> same result. For instance his is always where as mine is frequent but not
always.
> After I got to finally see it first hand today, I had to start over 
> from Caller 1 5 times to get it to happen again.
>
> Caller 1 calls in and Person A answers. Caller 2 calls in and Person B 
> answers. Person B puts caller 2 on hold and audio drops on Caller 1. 
> So Person A can hear caller 1 but caller 1 cannot hear Person A.
>
> This happens more often when Call 1 is on the handset and Call 2 is on 
> the portable or vis a vi, but this is not always the case. It does 
> happen to 1 set only but just less frequent.
>
> I have tried carrierinvite=yes and no but this does not change the issue.
> The phones are behind a router but the external IP of the router is on 
> the same network as the * box.
>
> Thanks!
>
> Curt
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Monday, November 06, 2006 6:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Question on Aastra phones and Astrisk
>
> Hi,
>Some odd behaviour here.  A friend and I were talking tonight,
> and it seems we have both seen the same problem.   We are both using
> aastra phones (I am using 9113is).We have a connection to and from
> providers via SIP and IAX.When I place a call on the local hold of
> the phone, and then pick them back up I can hear them, but they can
> not hear me.However, if I park the call, and then pick it up
> again, the audio is fine.
>   Tonight I tried placing a call on hold using a Sipura/Linksys 
> ATA (that is just hitting 'flash', which basically puts the call on
> local hold and starts music).The problem did not manifest itself.
>
> Has anyone else had this issue?  Do you have a fix for it?  It is an 
> astrisk issue or an aastra issue?
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RE: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

[shadowym]

Steve,

I am running EXACTLY the same software as you.  SMP kernel and all.  Are you
using the IDENTICAL non-SMP kernel verision# ?  No crashes at all here for
the 2 weeks it has been up but we don't use meetme. 

-Original Message-
From: RR [mailto:[EMAIL PROTECTED] 
Sent: Monday, November 06, 2006 11:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

On 11/2/06, Steve Edwards <[EMAIL PROTECTED]> wrote:
> I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application 
> is mostly meetme conferences being created and closed all day long. 
> Peak load is around 200 SIP calls.
>
> I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I 
> haven't had a crash since. Meetme does not play well with SMP.

HI Steve,

I have constantly got conflicting reports about meetme and can't really make
up my mind to actually put meetme into service till I find something better
or just stick with meetme and be happy? I like the features it has but
performance wise I have heard all sorts of things, yours being the most
positive so far. so just wondering if I can learn something from you.

So, is there anything special you've done in terms of configs, modules, OS
tweaking/tuning and the like, in other words, anything over and above simply
installing OS and (*) with meetme for the system mentioned above? Have you
standardised codecs across the board to minimise translation overhead? If
so, then what codec are you using?
Are all your users on IP or some can come through the PSTN via DIDs etc?

Thanks
Ranj


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RE: [asterisk-users] Queues and multiple lines

[Wes Baehr]

The default queue configuration would achieve this. Based on your queue
calling method (ringall, roundrobin, etc), all the agents would be able to
receive the 2nd call, and whoever answers it first gets it.

Wes Baehr

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Sampson
> Sent: Tuesday, November 07, 2006 1:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Queues and multiple lines
> 
> Say I have agents using a softphone like eyebeam that has 6 lines. They
> log in to the queue. Say there are 3 agents in my queue. 3 calls come in
> and all three agents are on a call. Now a fourth call comes in. Is it
> possible to have it setup so that the 4 call rings on line 2 of one of
> my agents, if they don't get it within the time limit it rings on line 2
> of another agent and so on. An agent can then put their current call on
> hold and go to the new call, say something like "thanks for calling
> please hold", then go back to their first call, finish it up and then go
> back to the second call. I hope that made sense. I'm sure there is a way
> to get it done, but how flexible is the current queue system in Asterisk
> with stuff like this?
> 
> --
> Michael Sampson
> Information Systems Manager
> Customer Contact Services
> [EMAIL PROTECTED]
> 952-936-4000
> 
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[asterisk-users] Pressing "*" makes Asterisk destroy my call

[Stefan Agethen]

I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN 
Cards, if i press in a call the "*" Asterisk, Asterisk destroys the call 
not, Asterisk lets him hang and do nothing, if i hangup, Asterisk tell 
me in the warnings-log that the bridging was not successfull ?!


If have disabled the function to hangup in the features.conf, but the 
key is still available, can someone explain me whats going on there ?


Stefan
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re: [asterisk-users] connect Sipura with Asterisk - both behind NAT

[Alyed Tzompa]

		If your Asterisk is behind a NAT you can use externip=x.x.x.x (sip.conf)
If your Sipura is behind a NAT you can use nat= yes (sip.conf)
		Btu I'm really afraid that unless you use a SIP
proxy (e.g. Portaone) you won't be able to succesfully connect both
elements if they both are behind NATs. It is just because of how SIP and NAT work together.
		
		
Alyed 
		
		
		
Return-Path: <[EMAIL PROTECTED]> Tue Nov 07 09:16:25 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;   Tue, 7 Nov 2006 09:16:25 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])
		
		Does anybody have a good link how to connect Sipura with Asteriks, bothbehind NAT?  I'm using FWD but their connection is like a weather(especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good linkexplaining how to setup Linux server-- #Joseph___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

[JR Richardson]

Hi All,

I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:

asterisk sip >< sip TNT pri >< pri asterisk

The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1.  I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the SIP Asterisk server. I
have tried many variations of using sip options insecure,
autocreatepeer, permit/deny, host, user, etc but can't seem to get
asterisk to accept an unauthenticated call from the TNT using SIP.  I
keep getting SIP/2.0 407 Proxy Authentication Required.  I know others
have done this, but with older Asterisk versions, I'm wondering what
versions of Asterisk are known to work with the MAX TNT and with what
version of the TNT?

I'm confident this is an asterisk issue, with insecure=very, I should
be able to pass calls to asterisk without trying to authenticate it
first.  But this is not happening.

Here is a debug of a call and a snip from my sip.conf:

sip.conf

[maxtnt]
type=friend
host=10.10.14.131
insecure=very
dtmfmode=inband
callerid="MaxTNT" 
context=trunktntin
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw

debug

lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
Remote-Party-Id: "NO CID NAME"
;screen=no;id-type=subscriber;party=calling;privacy=off
i: [EMAIL PROTECTED]
CSeq: 639089 INVITE
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
m: 
k: replaces
c: application/sdp
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
l: 232

v=0
o=t1gw01 531756636 531756636 IN IP4 10.10.14.131
s=Session SDP
c=IN IP4 10.10.14.131
t=0 0
m=audio 40198 RTP/AVP 0 96
a=silenceSupp:on
a=ecan:b on g168
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=rtpmap:0 PCMU/8000

--- (16 headers 11 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 10.10.14.131 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.10.14.131:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131
From: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
To: ;tag=as41f8454e
Call-ID: [EMAIL PROTECTED]
CSeq: 639089 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a"
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Found user '1239'
lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
t:   ;tag=as41f8454e
f: "NO CID NAME"
;tag=5fe9f589-1fb1f65c-830e0a0a
i: [EMAIL PROTECTED]
CSeq: 639089 ACK
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
User-Agent: Lucent-Universal-Gateway
l: 0


Any guidance will be much appreciated.

Thanks.

JR

--
JR Richardson
Engineering for the Masses
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RE: [asterisk-users] "Sticky" Polycom 501 keys and handset

[Bill Gibbs]

I think in the features you can completely
wipe the sip image

 

Menus

Settings

Advanced

Admin Settings

Reset to default

Then format the file system

 

Bill

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, November 07, 2006
1:49 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users]
"Sticky" Polycom 501 keys and handset



 

Any hints on downgrading?  I placed
the old SIP 1.6.7 on the right folder, but my phone wont pick it up and install
it.  It must be thinking "this is an old version, ignore" or
something

 

I`ve never downgraded a phone, I tend to
like upgrading more :-)

 

Mike



 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick Smith
Sent: November 7, 2006 11:28 AM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users]
"Sticky" Polycom 501 keys and handset

I had this EXACT
same problem, and 2.0.x is the problem according to Polycom Tech Support.

 

I had such a hard
time explaining the problem, too…

 

Downgraded to 1.6.7
and all worked well again.  Polycom says if you’re using Asterisk,
don’t

go past 1.6.7 until
they say to.

 





From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, November 07, 2006
11:02 AM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users]
"Sticky" Polycom 501 keys and handset





 



Hi,





 





I've recently bought new Polycom 501
phones, upgraded to bootrom 3.2.2 and SIP 2.0.1.   I just noticed
something, which I first blamed on Asterisk and NATs (a 2 second silence at the
beginning of a call).  Something I've noticed also on my old phone
(which is having the same problem now, but its also been upgraded).





 





My keys are sticky.  Simple as
that.  Sometimes I press a number and the key comes up (the hardware seems
fine) but the phone produces this lng tone as if I had pressed the key for
3 seconds.  Even the receiver is sticky, giving my dialtone when I lift it
only 1-2 seconds after I lift the handset.  It simply looks like the
phone can't keep up, like a sluggish computer.





 





Anybody has ever seem this? 
I'd like to downgrde to SIP 1.6.7 to see if the new sip app was the
problem.  How can I do that?  I've placed the old sip.ld file where I
had to, but the phone wont pick it up.  





 





Short of that, can somebody point me
to the newest firmware (2.0.2) to see if that would help?





 





Mike 










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RE: [asterisk-users] Re: Port Range

[shadowym]

Also, change the port range to 1-10xxx in /etc/asterisk/rtp.conf and
match it up in your port forward on your firewall instead of 1-2
which is far more anyone is likely to need. 

-Original Message-
From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED] 
Sent: Monday, November 06, 2006 9:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Port Range

Webmin uses UDP?  Audio is generally RTP over UDP.

Tom Vile wrote:
> That probably because you are using Webmin.  Just change the port 
> Webmin listens on instead, I use 9000.
> 
> On 11/6/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
>>
>> I'll keep that in mind for future. I read about using 10001 as start 
>> port on Nerd Vittles website.
>>
>> Is there some good material online to read more about RTP, SIP, RTCP 
>> and UTP?


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[asterisk-users] How is ANI "Usually" sent on an ISDN PRI?

[Steve Totaro]

I am getting ANI over an ISDN PRI T1 and they are sending *ANI*DNIS* as 
the "EXTEN"   I am under the impression this is an old way of doing it 
on inband circuits such as E&M wink but not used on a PRI.


Previously I had a T1 PRI from a different carrier and there was a 
special field for ANI that could be viewed while watching a "pri debug 
span" on the CLI.


I am trying to explain that this is the preferred method but am at a 
loss to show them examples or speak the right terminology.


Can someone post some debug output with the ANI information or let me 
know the the actual terms.  I know I am right but I am at a loss in 
trying to explain this to them.


Thanks,
Steve
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[asterisk-users] hicecaller ID

[Nik Engel]

Hi all !

I have a question regarding flexible callerid setting
using the misdn

I want to acheive the following:

when starting a call with 0 I want to display CALLERID (which is setup
right now) but when I start the call with 9 I want the callerid to be
surpressed.

How can this be done?

nik




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[asterisk-users] loosening voicemail file permissions for msg????.txt and msg????.wav

[Scott Keagy]

HI folks,

 

I figured out where in the source code to hack the .wav file
permissions which were set too restrictive for me, but I can’t figure out
how to do the same for the .txt file.

 

Looks like the voicemail.c file sets it nicely for
asterisk1.4beta3 using a #define statement early on, but msg.txt  comes
out with permissions 0600 and there are no umask entries that affect how
asterisk is started (if anything it would be 022).

 

I’ve grepped through the entire source tree from the
expanded tarball, and changed every place where it says 0600 to 0666 and
recompiled, but still no luck. I’ve grepped for umask entries like 077
that might cause the problem, but again no luck.

 

My work-around right now is a cron-job that chmods these
directories every minute, but this is ridiculous.

 

Anyone solved this?

 

Thanks,

Scott

 






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RE: [asterisk-users] "Sticky" Polycom 501 keys and handset

[Mike]

Disregard my previous message, I succeeded in downgrading 
my phones.  And it worked, thanks Rick for the info.  Is there any 
Polycom-specific mailing list I should be on to be aware of stuff like 
that?
 
Also, would you know how to check the version of sip.ld 
remotely? I know how to reboot remotely, and I did for a few phones, but my 
paranoid self would like to double check and see if the sip.ld 1.6.7 
re-installed ok by checking the current version.  Is that even 
possible?
 
Mike

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Rick 
  SmithSent: November 7, 2006 11:28 AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE: 
  [asterisk-users] "Sticky" Polycom 501 keys and handset
  
  
  I 
  had this EXACT same problem, and 2.0.x is the problem according to Polycom 
  Tech Support.
   
  I 
  had such a hard time explaining the problem, too…
   
  Downgraded 
  to 1.6.7 and all worked well again.  Polycom says if you’re using 
  Asterisk, don’t
  go 
  past 1.6.7 until they say to.
   
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  MikeSent: Tuesday, November 07, 2006 11:02 AMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  [asterisk-users] "Sticky" Polycom 501 keys and 
  handset
   
  
  Hi,
  
   
  
  I've recently 
  bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 
  2.0.1.   I just noticed something, which I first blamed on Asterisk 
  and NATs (a 2 second silence at the beginning of a call).  Something 
  I've noticed also on my old phone (which is having the same problem now, 
  but its also been upgraded).
  
   
  
  My keys are 
  sticky.  Simple as that.  Sometimes I press a number and the key 
  comes up (the hardware seems fine) but the phone produces this lng tone as 
  if I had pressed the key for 3 seconds.  Even the receiver is sticky, 
  giving my dialtone when I lift it only 1-2 seconds after I lift the 
  handset.  It simply looks like the phone can't keep up, like a 
  sluggish computer.
  
   
  
  Anybody has ever 
  seem this?  I'd like to downgrde to SIP 1.6.7 to see if the new sip app 
  was the problem.  How can I do that?  I've placed the old sip.ld 
  file where I had to, but the phone wont pick it up.  
  
  
   
  
  Short of that, can 
  somebody point me to the newest firmware (2.0.2) to see if that would 
  help?
  
   
  
  Mike 
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Re: [asterisk-users] connect Sipura with Asterisk - both behind NAT

[Juan Manuel Sá]

Try this site:
http://forum.voxilla.com/asterisk-users-group/sipura-asterisk-setup-14252-3.html

Juan Manuel

On 11/7/06, Joseph <[EMAIL PROTECTED]> wrote:

Does anybody have a good link how to connect Sipura with Asteriks, both
behind NAT?  I'm using FWD but their connection is like a weather
(especially IAX), I need something more reliable.

I was thinking of using stun and/or proxy but can not find any good link
explaining how to setup Linux server

--
#Joseph
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--
Juan Manuel Sá
(54.11) 15-4429-6560
(54.11) 4824-9048 (int 218)
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RE: [asterisk-users] "Sticky" Polycom 501 keys and handset

[Mike]

Any hints on downgrading?  I placed the old SIP 1.6.7 
on the right folder, but my phone wont pick it up and install it.  It must 
be thinking "this is an old version, ignore" or 
something
 
I`ve never downgraded a phone, I tend to like upgrading 
more :-)
 
Mike

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Rick 
  SmithSent: November 7, 2006 11:28 AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE: 
  [asterisk-users] "Sticky" Polycom 501 keys and handset
  
  
  I 
  had this EXACT same problem, and 2.0.x is the problem according to Polycom 
  Tech Support.
   
  I 
  had such a hard time explaining the problem, too…
   
  Downgraded 
  to 1.6.7 and all worked well again.  Polycom says if you’re using 
  Asterisk, don’t
  go 
  past 1.6.7 until they say to.
   
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  MikeSent: Tuesday, November 07, 2006 11:02 AMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  [asterisk-users] "Sticky" Polycom 501 keys and 
  handset
   
  
  Hi,
  
   
  
  I've recently 
  bought new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 
  2.0.1.   I just noticed something, which I first blamed on Asterisk 
  and NATs (a 2 second silence at the beginning of a call).  Something 
  I've noticed also on my old phone (which is having the same problem now, 
  but its also been upgraded).
  
   
  
  My keys are 
  sticky.  Simple as that.  Sometimes I press a number and the key 
  comes up (the hardware seems fine) but the phone produces this lng tone as 
  if I had pressed the key for 3 seconds.  Even the receiver is sticky, 
  giving my dialtone when I lift it only 1-2 seconds after I lift the 
  handset.  It simply looks like the phone can't keep up, like a 
  sluggish computer.
  
   
  
  Anybody has ever 
  seem this?  I'd like to downgrde to SIP 1.6.7 to see if the new sip app 
  was the problem.  How can I do that?  I've placed the old sip.ld 
  file where I had to, but the phone wont pick it up.  
  
  
   
  
  Short of that, can 
  somebody point me to the newest firmware (2.0.2) to see if that would 
  help?
  
   
  
  Mike 
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Re: [asterisk-users] capiAnswerFax

[Michiel van Baak]

On 15:34, Tue 07 Nov 06, Pedro Silva wrote:
> Excellent, Michiel! This works :)
> You know what kind of file it is created (SFF)?
> Can you send to me the example faxreceive.php?
> Thanks and best regards!
> PS.

Hi,

Glad it worked. The generated file is a 'Structured Fax
File'
The faxreceive.php is not that important in this case. All
it does is lookup customer that sends the fax in our CRM
application and moves the faxfile to the binary datastore of
our CRM application.
But if you're still interested here is the subversion view
of it:
http://covide.svn.sourceforge.net/viewvc/covide/trunk/classes/voip/agi_scripts/mysql/faxreceive.php?revision=1&view=markup

What's more interesting for you I think is how you can
convert this file into something more common.

The code is here:
http://covide.svn.sourceforge.net/viewvc/covide/trunk/classes/voip/data.php?revision=1&view=markup
line 80 and below.

Basically you can run the following commandline tools on the
file to create a .pdf
sfftobmp -t  -o 
tiff2pdf -o  

Good luck
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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[asterisk-users] Queues and multiple lines

[Michael Sampson]

Say I have agents using a softphone like eyebeam that has 6 lines. They 
log in to the queue. Say there are 3 agents in my queue. 3 calls come in 
and all three agents are on a call. Now a fourth call comes in. Is it 
possible to have it setup so that the 4 call rings on line 2 of one of 
my agents, if they don't get it within the time limit it rings on line 2 
of another agent and so on. An agent can then put their current call on 
hold and go to the new call, say something like "thanks for calling 
please hold", then go back to their first call, finish it up and then go 
back to the second call. I hope that made sense. I'm sure there is a way 
to get it done, but how flexible is the current queue system in Asterisk 
with stuff like this?


--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

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Re: [asterisk-users] g729

[Tzafrir Cohen]

On Wed, Nov 08, 2006 at 08:06:21AM -0800, Khaled wrote:
> Does digium have a g723 codec can work  pass thru mode 

Digium does not provide a g723 codec. However to work in pass-through
mode you don't need a codec. A codec is used when transcoding the
stream.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] g729

[Joshua Colp]

Khaled wrote:



Does digium have a g723 codec can work  pass thru mode



You don't need a codec loaded to do passthru, Asterisk just needs to 
know about the codec (which it does in the case of G723.1). If things 
are properly configured and codec negotiation happens the right way, it 
should just work.


--
Joshua Colp
Software Developer
Digium, Inc.
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Re: [asterisk-users] How do I make this stop? (Bridging of IAX channels?)

[Joshua Colp]

Matt wrote:

   -- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21

I want everything to stay in the VoIP server rather then briding.  I
have notransfer=yes on, but it still seems to bridge the call
natively..  can I keep the RTP stream on the asterisk server some how?


Asterisk is still going to try to native bridge the two channels. Once 
this occurs chan_iax2 is going to notice that you don't want a native 
transfer to happen and not do it.


--
Joshua Colp
Software Developer
Digium, Inc.
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RE: [asterisk-users] "Sticky" Polycom 501 keys and handset

[Rick Smith]

I had this EXACT same problem, and 2.0.x is the problem
according to Polycom Tech Support.

 

I had such a hard time explaining the problem, too…

 

Downgraded to 1.6.7 and all worked well again.  Polycom says if
you’re using Asterisk, don’t

go past 1.6.7 until they say to.

 





From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, November 07, 2006 11:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] "Sticky" Polycom 501 keys and
handset





 



Hi,





 





I've recently bought new Polycom 501 phones,
upgraded to bootrom 3.2.2 and SIP 2.0.1.   I just noticed something,
which I first blamed on Asterisk and NATs (a 2 second silence at the beginning
of a call).  Something I've noticed also on my old phone (which is
having the same problem now, but its also been upgraded).





 





My keys are sticky.  Simple as
that.  Sometimes I press a number and the key comes up (the hardware seems
fine) but the phone produces this lng tone as if I had pressed the key for
3 seconds.  Even the receiver is sticky, giving my dialtone when I lift it
only 1-2 seconds after I lift the handset.  It simply looks like the
phone can't keep up, like a sluggish computer.





 





Anybody has ever seem this?  I'd like to
downgrde to SIP 1.6.7 to see if the new sip app was the problem.  How can
I do that?  I've placed the old sip.ld file where I had to, but the phone
wont pick it up.  





 





Short of that, can somebody point me to the
newest firmware (2.0.2) to see if that would help?





 





Mike 








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[asterisk-users] Grandstream TFTP system wide settings

[Zeeshan Zakaria]

Hi,
 
Aastra IP Phones have two configuration files on TFTP, aastra.cfg and .cfg. Both are in text format, which makes editing easy. And aastra.cfg has system wide settings and .cfg has settings for each indivifual phones. This makes it really easy to change the global parameters system wide by changing only one 
aastra.cfg file.
 
On the other hand, as I could understand, for Grandstream TFTP setup, each phones needs a separate file, which has to be edited and then converted to its own format usint ./encode.sh. There is no such file which would carry global settings for all the phones on a system. Changing 10s of configuration files for one small little thing, like daylight saving = 0, and then converting all of them to its own format is not a good way of dealing with many phones.

 
Is there a quicker way to change settings for all Grandstream phones, is there any one file which can act as a global configuration file without changing each phones phone specific settings? And can't it be simply done by text editing, without the need to convert each file to cfg format?

 
-- Zeeshan A Zakaria 
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[asterisk-users] my future count on this please help

[Khaled]

 

Dear

 

How can I charge the incoming call to the destination call
,using a2billing 

I used to make setaccount but it didn’t work such a
loopback detected am using
context=default for incoming calls 

 

I

 

 

Please help 

Thanks 

 






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[asterisk-users] g729

[Khaled]

Does digium have a g723 codec can work  pass thru mode 

 






*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.

If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.
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Re: [asterisk-users] operator console

[Stephen Wingfield]

Andres,

The Bicom Systems Operator Panel is probably what you are looking for. OPCOM

http://www.bicomsystems.com/docs/opcom/1.0/html/

This is included with every copy of PBXware and is fully supported.
If you care to register you may order a trial of PBXware with our SOHO.

Regards
Steve
steve 'at' bicomsystems 'dot' com



- Original Message - 
From: "Andres Paglayan" <>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, October 30, 2006 5:27 PM
Subject: [asterisk-users] operator console



Hi,

My users are currently using an operator console interface like this:
see it at: http://www.whssf.org/interface.jpg

which came with a Praxon PDX we got about 5 years ago, which is now 
unsupported,
it works very good and converts any analog phone plugged into the  system 
into a powerful console,

(provided you have a computer next to it)
you just provide the box ip, user login, user pass, and extension,  and 
voila.


I'll be switching the company's phone system to Asterisk.

I know * is way much more flexible and rich featured than the box we 
currently have,


...but I'll need to give the users a good mean to "see"
what's going on,
who is busy,
easy transfer with click here and there,
easy conference,
easy queue handler,
easy way to see/use many lines at the same time

is there any "best" console they can use?

I don't mind using a commercial product,
if the only part we have to pay for is the gui,
besides, we will buying the enterprise * version

Thanks a bunch,

Andres

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Re: [asterisk-users] Audiocodes MP-114 noise

[Jessee J Holmes]

Jason,First, before you start reading, get to the latest firmware from Audiocodes (MP118_SIP_F4.80A.034.004.cmp), there have been significant echo improvements in this version.After many days of working with Audiocodes on this problem and much time spent here by multiple technicians trying to reproduce and resolve this issue; this morning, Atacomm received an email from Audiocodes with a full explanation to this now confirmed issue with all MP-11x units. Atacomm will immediately begin work on a KB article within our website that confirms this issue and outlines the manufacturer recommended steps to resolve this problem.Apparently, there have been some changes with the MP-11x's that can negatively affect line noise and echo.  Below are some steps which can help to correct these problems:1. The new design did away with the Coefficent file.  Audiocodes, now instead, introduced a configurable parameter called countrycoefficient. This parameter can be adjusted to a specific country based on known configurations.  For the most part this should work.  70(USA) is the default value.  More can be found in the User’s manual.2.  In just about every case, an FXO is added to a Pre-existing PBX or CO line, you can expect echo. This comes from the fact that delay (IP Network) is being introduced, and what used to be Side tone is now delayed so much it is echo. Just about every difference on the line that can be heard between the pre fxo and post fxo installation can be traced to echo, or line quality issues.3.  Going forward, Audiocodes would like to suggest that when installing the product do the following:A) Make sure the Line coming from the PBX or CO is a Loop Start line. Ground start is not supported on the MP-11x series of gateways. (The M1K FXO will in 5.0)B) Check that the Line can deliver for a 600 Ohm Impedance line-52 to -24 V of Off Hook Voltage-15 to -6 V of  On Hook Voltage20 to 35 ma of loop current.If you know the line is not 600 Ohm, please gather metrics on the line, and the make and model of the PBX or switch it is attached too, plus country of origin. If it is not from the USA, please look up the country of origin and then find the CountryCoefficient to match this. Load the .ini file to the board with this setting and reset.  Make sure the Gateway has a firmware version of 4.60.035 or higher or 4.80.030 or higher.C) Put the device on the network with Voice Volume set to 0 and input gain set to 0. Make calls, if there is no issue, you can stop here.  However, Echo is still expected most of the time.D) The echo should be heard by the IP side participant as their voice is reflected back.  If this is the case, then what needs to be done is to lower the voicevolume (IP—TEL). This way the speaker’s reflected voice will comeback low enough for the ECAN to cancel it out (-6 is usually recommended as the value to plug in here). A little experimentation is needed as the loss for all lines will vary based on length from the CO. Echo is usually taken care of in this manner.E) The incoming speaker from the PSTN’s voice seems low, set InputGainLocation =1, and then slowly increment the Input Gain Parameter(TelàIP) to adjust for this. In past releases (see the note about loads above), the input gain was always applied prior to the ECAN which had the effect of amplifying the returned echo and noise on the line causing crosstalk and clipping issues. This is no longer the case.If the above does not resolve the issues, then you need to go ahead and collect DSP, Ethereal and Syslog traces along with the board.ini, these are to be sent to your support agent, who will then send these to Audiocodes for their engineers to evaluate.  This should not happen often.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Nov 3, 2006, at 12:14 AM, Jason Kim wrote:Jessee,I tried many combinations of "Voice Volume", "InputGain" and packetization time , but it's noisy steel.I'm using G.711A-law and packetization time is 20ms.It can be impedance mismatch problem but i cannotadjust impedance of FXO port of MP-114.___
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Re: [asterisk-users] Reading Voicemail Config from MySQL [+ ODBC]

[RR]

On 11/6/06, Mosiuoa Tsietsi <[EMAIL PROTECTED]> wrote:

Hi,

After some more searching I decided to try USING unix ODBC for the
connection.  I have both the unixODBC and unixODBC-devel packages on my
fedora box:

[EMAIL PROTECTED] /]# rpm -qa | grep -i unixodbc
unixODBC-2.2.11-7.1
unixODBC-devel-2.2.11-7.1

Here are my odbcinsi.ini and odbc.ini files respectively:

[MySQL]
Description = ODBC for MySQL
Driver  = /usr/lib/libmyodbc.so
Setup   = /usr/lib/libodbcmyS.so
FileUsage   = 1

---
[MYSQL-asterisk]
Driver = MySQL
Description = Data source for dynamic asterisk voicemail configuration
Trace = Yes
TraceFile = stderr
SERVER = localhost
USER = root
PASSWORD = rootroot9
PORT = 3306
DATABASE = asterisk
-

Below are my res_odbc.conf and extconfig.conf files for supplying
details of the DSN name and and database/table for asterisk

[mysql1]
enabled => yes
dsn => MySQL-asterisk
username => root
password => ***
pre-connect => yes

---
[settings]
voicemail => odbc,mysql1,users
---
I am able to execute:
[EMAIL PROTECTED] /]# isql -v MySQL-asterisk
+---+
| Connected!|
|   |
| sql-statement |
| help [tablename]  |
| quit  |
|   |
+---+
SQL>

which shows I can connect to the database on the command line using my
DSN name.

In the asterisk CLI however, the command:

asterisk*CLI> odbc show
No such command 'odbc' (type 'help' for help)

fails which is supposed to show connections to MySQL from the CLI.  ANd
lastly the command:

asterisk*CLI> realtime load voicemail mailbox 7521
No rows found matching search criteria.
Nov  6 00:33:10 WARNING[2965]: config.c:920 find_engine: Realtime
mapping for 'voicemail' found to engine 'odbc', but the engine is not
available

also fails.  Where are I going wrong?
Thanks.



Mate, doesn't sound like you have the res_odbc.so module loaded. Make
sure in /etc/asterisk/modules.conf you have a load => res_odbc.so or
on the CLI type load res_odbc.so and then give it a whirl.
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[asterisk-users] Generating Recall/Flash using Zaptel

[Dululu Ululu]

HiI'm trying to generate a Recall/Flash on an FXO (TDM400) connected to a PABX and failing at the moment. I was using the Flash application, which seems to generate a hook flash as opposed to a Recall. Debugging the Zap channel output using a second FXS channel gives me the following
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1]<< [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/1-1]when I press Recall on the type of phone I'm trying to mimic and<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1]
<< [ TYPE: Control (4) SUBCLASS: Flash (9) ] [Zap/1-1]when I use the Flash application.  I have 3 questions which I'd greatly appreciate answers to1) Is it possible to generate the 
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1]only using a Zap channel? 2) If it is, is there an easy way to do this, or do I have to delve into zaptel / asterisk code?3) If I have to delve into asterisk code, which areas should I be looking at? As a newbie to the whole zaptel architecture and 
asterisk code base who is on a rather tight schedule if I have to go this route then any pointers as to where I should be looking and what I should be doing would be hugely appreciatedI've searched high and low and can't seem to find answers to any of these questions.
ThanksDululu
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[asterisk-users] connect Sipura with Asterisk - both behind NAT

[Joseph]

Does anybody have a good link how to connect Sipura with Asteriks, both
behind NAT?  I'm using FWD but their connection is like a weather
(especially IAX), I need something more reliable. 

I was thinking of using stun and/or proxy but can not find any good link
explaining how to setup Linux server

-- 
#Joseph
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[asterisk-users] "Sticky" Polycom 501 keys and handset

[Mike]

Hi,
 
I've recently bought 
new Polycom 501 phones, upgraded to bootrom 3.2.2 and SIP 2.0.1.   I 
just noticed something, which I first blamed on Asterisk and NATs (a 2 second 
silence at the beginning of a call).  Something I've noticed also on 
my old phone (which is having the same problem now, but its also been 
upgraded).
 
My keys are 
sticky.  Simple as that.  Sometimes I press a number and the key comes 
up (the hardware seems fine) but the phone produces this lng tone as if I 
had pressed the key for 3 seconds.  Even the receiver is sticky, giving my 
dialtone when I lift it only 1-2 seconds after I lift the handset.  It 
simply looks like the phone can't keep up, like a 
sluggish computer.
 
Anybody has ever 
seem this?  I'd like to downgrde to SIP 1.6.7 to see if the new sip app was 
the problem.  How can I do that?  I've placed the old sip.ld file 
where I had to, but the phone wont pick it up.  
 
Short of that, can 
somebody point me to the newest firmware (2.0.2) to see if that would 
help?
 
Mike 
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[asterisk-users] hicecaller ID

[Nik Engel]

Hi all !

I have a question regarding flexible callerid setting
using the misdn

I want to acheive the following:

when starting a call with 0 I want to display CALLERID (which is setup
right now) but when I start the call with 9 I want the callerid to be
surpressed.

How can this be done?

nik



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Re: [asterisk-users] capiAnswerFax

[Pedro Silva]

Excellent, Michiel! This works :)
You know what kind of file it is created (SFF)?
Can you send to me the example faxreceive.php?
Thanks and best regards!
PS.

2006/11/7, Michiel van Baak <[EMAIL PROTECTED]>:

On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
> Hello,
>
> Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
> I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
> it give me errors...
> Also i tried define one extension for fax receptions but this dont works:
> exten => 1,1,Goto(handle_fax,s,1)
> exten => fax,1,Goto(handle_fax,s,1)
>
> [handle_fax]
> exten => s,1,capiAnswerFax(/tmp/${UNIQUEID})
> exten => s,2,Hangup()
>
> With this code, a fax call to DID 1 must be attended and the fax
> stored in /tmp, right?
> This not works... :(
>
> Thanks for any kind of possible help...
> PS.

Hi,

The chan_capi you mention already has fax support.
Here is the handle_fax context I use with the latest
released chan_capi-cm

[handle_fax]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID})
exten => s,n,capicommand(receivefax|${FAXFILE})
exten => h,1,DeadAgi(faxreceive.php|${FAXFILE})

Good luck
--

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [asterisk-users] Snom 360 flickering screen

[Dovid B]

I had a problem with snom where the screen went completly blank. Snom told 
me there was an issue where that the cable going from the phone board to the 
screen would fall out. I opend the phone and sliped it back in.


- Original Message - 
From: "Nick Hoffman" <[EMAIL PROTECTED]>

To: "asterisk-users Mailing List" 
Sent: Tuesday, November 07, 2006 10:53 AM
Subject: [asterisk-users] Snom 360 flickering screen



Hi guys. I just bought and configured a Snom 360 and have noticed that the
LCD is constantly flickering at a rate of ~10-15Hz (that's a guess).
Either way, it's very distracting. Has anyone else encountered this
before? Any solutions?

Cheers,
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any
use of the email.  We do not waive any privilege, confidentiality or
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[asterisk-users] failed to authenticate on invite

[Damon Estep]

I have 2 asterisk boxes connected via SIP

 

box 1 sip peer connected
to box 2 (ip addresses intentionally removed)

 

[ast20]

type=friend

host=x.x.x.20

insecure=very

context=subscriber

dtmfmode=inband

qualify=no

canreinvite=no

disallow=all

allow=ulaw

 

box 2 sip peer connected
to box 1

 

[sbb19]

type=friend

host=64.1.8.19

insecure=very

context=inbound

dtmfmode=inband

qualify=yes

canreinvite=no

disallow=all

allow=ulaw

 

I then have 2 UAs registed on box 1, both have identical
configs with the exception of username, but one is a Polycom IP501 and the
other is a Linksys PAP2

 

The IP 501 can call to box 2 with no issues, also calls
originated on a PRI connected to box 1 connect to box 2 with no issues.

 

The Linksys UA can not call box 2, here is the error
(numbers intentionally removed);

 

-- Executing dial("SIP/##0850-b6669f58",
"SIP/[EMAIL PROTECTED]")

    -- Called [EMAIL PROTECTED]

Nov  7 07:20:45 NOTICE[21059]:
chan_sip.c:9709 handle_response_invite: Failed to authenticate on INVITE to
'"name removed" ;tag=as38826922'

    -- SIP/ast20-09c8b110
is circuit-busy

  == Everyone is busy/congested
at this time (1:0/1/0)

 

I have looked at sip debugs from
both scenarios, and the invites from box 1 to box 2 look nearly identical, box
2 never shows the call when it fails.

 

I am assuming that there is
something that needs to be changed on the ATA or peer config to get it to be
able to call via box1 to box2 without requiring authentication, but can not
figure out what.

 

Any ideas?

 

 

 






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Re: [asterisk-users] capiAnswerFax

[Armin Schindler]

On Tue, 7 Nov 2006, Pedro Silva wrote:
> Hello,
> 
> Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?

This patch is not added to chan-capi.org, but receivefax and sendfax is 
available via capicommand(). Please see README of chan-capi 0.7.x package.

Armin

> I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
> it give me errors...
> Also i tried define one extension for fax receptions but this dont works:
> exten => 1,1,Goto(handle_fax,s,1)
> exten => fax,1,Goto(handle_fax,s,1)
> 
> [handle_fax]
> exten => s,1,capiAnswerFax(/tmp/${UNIQUEID})
> exten => s,2,Hangup()
> 
> With this code, a fax call to DID 1 must be attended and the fax
> stored in /tmp, right?
> This not works... :(
> 
> Thanks for any kind of possible help...
> PS.
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[asterisk-users] How do I make this stop? (Bridging of IAX channels?)

[Matt]

   -- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21

I want everything to stay in the VoIP server rather then briding.  I
have notransfer=yes on, but it still seems to bridge the call
natively..  can I keep the RTP stream on the asterisk server some how?
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Re: [asterisk-users] capiAnswerFax

[Michiel van Baak]

On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
> Hello,
> 
> Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
> I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
> it give me errors...
> Also i tried define one extension for fax receptions but this dont works:
> exten => 1,1,Goto(handle_fax,s,1)
> exten => fax,1,Goto(handle_fax,s,1)
> 
> [handle_fax]
> exten => s,1,capiAnswerFax(/tmp/${UNIQUEID})
> exten => s,2,Hangup()
> 
> With this code, a fax call to DID 1 must be attended and the fax
> stored in /tmp, right?
> This not works... :(
> 
> Thanks for any kind of possible help...
> PS.

Hi,

The chan_capi you mention already has fax support.
Here is the handle_fax context I use with the latest
released chan_capi-cm

[handle_fax]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID})
exten => s,n,capicommand(receivefax|${FAXFILE})
exten => h,1,DeadAgi(faxreceive.php|${FAXFILE})

Good luck
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [asterisk-users] Mapping CLI'S in Dialplan

[scott]

Hi

Thanks for reply but that wasnt quite what i was trying to explain :-)

Bascially a users callerID would be their extension. On offnet calls i needed 
to have the callerID reset as their DDI rather than their internal extension.

I endup using a mysql command in the dialplan to pull out a ddi based on the 
extension and the re-wrote me callerID when making offnet calls.

thanks
sp


-Original message-
From: Benjamin Jacob [EMAIL PROTECTED]
Date: Tue,  7 Nov 2006 05:24:21 -0600
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mapping CLI'S in Dialplan

>  Your offnet calls will be more than 4 digits, so use that to ur advantage.
>  so, for internal calls,
>  exten => _,1,Set(CALLERID(all)=Name ${CALLERID(num)})
>  or if u dont want to change the CLID at all.. dont do anything..
>  exten => _,1,NoOp(nothing)
>  
>  else, for all external calls(>4 digits)
>  exten => _X.,1,Set(CALLERID(num)=)
>  
>  
>  cheerz
>  - Ben.
>  
>  Scott Pinhorne wrote:
>  
>  > Hi All
>  >
>  >  
>  >
>  > I am not sure what I wish to do it possible but I would like to see if 
>  > you guys know any better.
>  >
>  >  
>  >
>  > I have a site who has the extensions: 1231, 1232. 1233, 1234
>  >
>  >  
>  >
>  > Each of these users can dial each other on the extension number an 
>  > also has an external CLI mapped to them.
>  >
>  > On all internal calls or calls to services such as call forwarding 
>  > their Caller ID is: Name 
>  >
>  >  
>  >
>  > What I would like to have happen is have the Caller ID changed to the 
>  > CLI only when they make an offnet call.
>  >
>  > So what I am saying is I need to match an extension to a CLI and reset 
>  > the Caller ID.
>  >
>  >  
>  >
>  > Many Thanks
>  >
>  > SP
>  >
>  >  
>  >
>  >  
>  >
>  >
>  >
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[asterisk-users] capiAnswerFax

[Pedro Silva]

Hello,

Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
exten => 1,1,Goto(handle_fax,s,1)
exten => fax,1,Goto(handle_fax,s,1)

[handle_fax]
exten => s,1,capiAnswerFax(/tmp/${UNIQUEID})
exten => s,2,Hangup()

With this code, a fax call to DID 1 must be attended and the fax
stored in /tmp, right?
This not works... :(

Thanks for any kind of possible help...
PS.
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Re: [asterisk-users] Follow Me problems

[David M. Zendzian]

I asked on freenode #asterisk a while ago about the followme and someone 
was nice enough to share a macro with me that I'm using (although I use 
voip outbound instead of zap, but it may be worth trying):


MYPHONE=IAX2/666
MYOUT=SIP/[EMAIL PROTECTED]
MYEXT=666

exten => 9,1,Macro(voicemail,${MYPHONE},${MYOUT},${MYEXT})

[macro-voicemail]
exten => s,1,Dial(${ARG1},25,r)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Setvar(NewCaller=${CALLERIDNUM})
exten => s-NOANSWER,2,Set(CALLERID(number)=${CALLERIDNUM})
exten => s-NOANSWER,3,Dial(${ARG2},15,r)
exten => s-NOANSWER,4,Set(CALLERID(number)=${NewCaller})
exten => s-NOANSWER,5,Voicemail(u${ARG3})
exten => s-NOANSWER,6,Hangup
exten => s-BUSY,1,Voicemail(b${ARG3})
exten => s-BUSY,2,Hangup
exten => _s-.,1,Goto(s-NOANSWER,1)

Time Bandit wrote:

Today we appear to have discovered our first bug.  We have an extension
setup to "followme" by ringing that extension + an external cell #
(ringall).  If nobody answers after 20 seconds the "destination if no
answer" is set to go to the extensions voicemail in the "followme" 
module.
The problem is it just keeps ringing forever.  If we delete the 
followme it

forwards to the voicemail as per the default SIP extension configuration
with voicemail enabled.

Anyone run into this?  Is there a workaround?  Any advice would be 
greatly

appreciated as always.

Our configuration is:
Supermicro Pentium D 2.66 Server with 2x512MB Memory
3ware 8006-2LP Hardware RAID 1
Sangoma A200D with 8fxo (latest firmware/drivers as of last week)
CentOS 4.4
Asterisk 1.2.13
Zaptel 1.2.10
FreePBX 2.1.3


When Asterisk dial the Cell phone, it goes out on the ZAP channel
(Sangoma A200D), so as soon as it hit that channel, the call is
considered answered even if the cell phone never actually pickup the
call. I didn't play with the "followme" module myself but that is what
I suspect is happening. Just watch the console and you should see
something like "Zap/1-1 answered ..."

hth
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Re: [asterisk-users] Dial plan Question

[Doug Crompton]

Thanks, that set off a light bulb In my spa3K my incoming dialplan was
set to  (S0<:405>)

Since this is a one FXO unit and my [from-pstn] will always be that line
can I make it generic and use the 's' extension as I described? If so what
would that spa3k dialplan be? just s0 ?

Doug

On Tue, 7 Nov 2006, Anselm Martin Hoffmeister wrote:

> Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
> > I am trying to do something that I see describe in a book and it is not
> > working
> >
> > In my sip.conf, I have in my [fxo] context=from-pstn
> >
> > I then have in extensions.conf
> >
> > [from-pstn]
> >
> > exten s,1,answer()
> > exten s,2,playback(blah)
> >
> > etc.
> >
> > It never answers but if I do this
> >
> > [from-pstn]
> >
> > exten _x.,1,answer()
> > exten _x.,2,playback(blah)
> >
> > it works.  Why does the 's' extension not work here?
>
> If "fxo" means your SIP provider, and you register with him, a specific
> extension will be called. Which one shall be called can be selected by
> the last parameter of the register statement, e.g.
>
> register => 075741:[EMAIL PROTECTED]:5060/492281234567
>
> will cause the incoming calls to appear in extension 492281234567.
>
> Comes in handy if you have several accounts with a single SIP provider:
> This way, you can simply distinguish the outward phone number for which
> the call came in.
>
> BR
> Anselm
>
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 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [asterisk-users] My Phone Review- Large Scale Corp Deployment.

[Andrew Joakimsen]

Honestly they are not that bad. When they first came out they were very buggy, but about 6 months ago I pulled one out and updated the firmware and actually over a few days was reliable.
On 11/7/06, RR <[EMAIL PROTECTED]> wrote:
On 11/2/06, Eddie Johnson Jr <[EMAIL PROTECTED]> wrote:> Hello Matthew,>> Did you test Snom or Sipura hard ip phones?  I was considering Budgetone for an office of
> 10 users.  After reading your testimonial I will have to re-think my selection.FWIW, after having played with 3-4 BudgeTone phones on 3-4 separateoccasions, out of which 2 actually just died the very next day (came
to work to find them, again on seperate occasions, with LCDs clearedout with greenish-blue tint, the speaker light lit and No Tone),that's when I concluded that the BudgeTone is surely Budget but NoTone!
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Re: [asterisk-users] Follow Me problems

[Time Bandit]

Today we appear to have discovered our first bug.  We have an extension
setup to "followme" by ringing that extension + an external cell #
(ringall).  If nobody answers after 20 seconds the "destination if no
answer" is set to go to the extensions voicemail in the "followme" module.
The problem is it just keeps ringing forever.  If we delete the followme it
forwards to the voicemail as per the default SIP extension configuration
with voicemail enabled.

Anyone run into this?  Is there a workaround?  Any advice would be greatly
appreciated as always.

Our configuration is:
Supermicro Pentium D 2.66 Server with 2x512MB Memory
3ware 8006-2LP Hardware RAID 1
Sangoma A200D with 8fxo (latest firmware/drivers as of last week)
CentOS 4.4
Asterisk 1.2.13
Zaptel 1.2.10
FreePBX 2.1.3


When Asterisk dial the Cell phone, it goes out on the ZAP channel
(Sangoma A200D), so as soon as it hit that channel, the call is
considered answered even if the cell phone never actually pickup the
call. I didn't play with the "followme" module myself but that is what
I suspect is happening. Just watch the console and you should see
something like "Zap/1-1 answered ..."

hth
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RE: [asterisk-users] Question on Aastra phones and Astrisk

[Gareth Owen]

Matt,

I haven't heard of this happening elsewhere, but I don't hear of every
issue with the phones.

It sounds to me that you've got more than enough information to raise a
case with our support team.


Gareth

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt
> Sent: 07 November, 2006 8:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk
> 
> *bump*  Anyone?
> 
> On 11/6/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:
> > I wanted to add what we have both seen on traffic captures.
> >
> > You see Caller 1's RTP stream. Call 2 comes in and you see the
creation
> of
> > its RTP stream. After Call 2 is put on hold the RTP stream from
Caller 1
> > disappears without a trace never to return and this is when the one
way
> > audio is happening.
> >
> > And I also wanted to add that I am running 1.4.0 firmware for this
> phone.
> >
> > Thanks again!
> >
> >
> >
> > -Original Message-
> > From: Curt Shaffer [mailto:[EMAIL PROTECTED]
> > Sent: Monday, November 06, 2006 6:58 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk
> >
> > I'm the friend mentioned here.
> >
> > I am using the Aastra 480i CT. It is SIP to my PBX and IAX
termination
> from
> > the PBX to my provider. My issue has a slight twist to it but the
same
> > result. For instance his is always where as mine is frequent but not
> always.
> > After I got to finally see it first hand today, I had to start over
from
> > Caller 1 5 times to get it to happen again.
> >
> > Caller 1 calls in and Person A answers. Caller 2 calls in and Person
B
> > answers. Person B puts caller 2 on hold and audio drops on Caller 1.
So
> > Person A can hear caller 1 but caller 1 cannot hear Person A.
> >
> > This happens more often when Call 1 is on the handset and Call 2 is
on
> the
> > portable or vis a vi, but this is not always the case. It does
happen to
> 1
> > set only but just less frequent.
> >
> > I have tried carrierinvite=yes and no but this does not change the
> issue.
> > The phones are behind a router but the external IP of the router is
on
> the
> > same network as the * box.
> >
> > Thanks!
> >
> > Curt
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: Monday, November 06, 2006 6:35 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Question on Aastra phones and Astrisk
> >
> > Hi,
> >Some odd behaviour here.  A friend and I were talking
tonight,
> > and it seems we have both seen the same problem.   We are both using
> > aastra phones (I am using 9113is).We have a connection to and
from
> > providers via SIP and IAX.When I place a call on the local hold
of
> > the phone, and then pick them back up I can hear them, but they can
> > not hear me.However, if I park the call, and then pick it up
> > again, the audio is fine.
> >   Tonight I tried placing a call on hold using a Sipura/Linksys
> > ATA (that is just hitting 'flash', which basically puts the call on
> > local hold and starts music).The problem did not manifest
itself.
> >
> > Has anyone else had this issue?  Do you have a fix for it?  It is an
> > astrisk issue or an aastra issue?
> > ___
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Re: [asterisk-users] Snom 360 flickering screen

[Chris Mazuc]

My company has had a few screens go out on us, but all of those were 
completely blank. I'm not sure if we just got a bad batch or what, but 
the Snom phones are usually a solid piece of hardware. I'd try to RMA it.


Nick Hoffman wrote:
Hi guys. I just bought and configured a Snom 360 and have noticed that the 
LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). 
Either way, it's very distracting. Has anyone else encountered this 
before? Any solutions?


Cheers,
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
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Re: [asterisk-users] Question on Aastra phones and Astrisk

[Matt]

*bump*  Anyone?

On 11/6/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:

I wanted to add what we have both seen on traffic captures.

You see Caller 1's RTP stream. Call 2 comes in and you see the creation of
its RTP stream. After Call 2 is put on hold the RTP stream from Caller 1
disappears without a trace never to return and this is when the one way
audio is happening.

And I also wanted to add that I am running 1.4.0 firmware for this phone.

Thanks again!



-Original Message-
From: Curt Shaffer [mailto:[EMAIL PROTECTED]
Sent: Monday, November 06, 2006 6:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk

I'm the friend mentioned here.

I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination from
the PBX to my provider. My issue has a slight twist to it but the same
result. For instance his is always where as mine is frequent but not always.
After I got to finally see it first hand today, I had to start over from
Caller 1 5 times to get it to happen again.

Caller 1 calls in and Person A answers. Caller 2 calls in and Person B
answers. Person B puts caller 2 on hold and audio drops on Caller 1. So
Person A can hear caller 1 but caller 1 cannot hear Person A.

This happens more often when Call 1 is on the handset and Call 2 is on the
portable or vis a vi, but this is not always the case. It does happen to 1
set only but just less frequent.

I have tried carrierinvite=yes and no but this does not change the issue.
The phones are behind a router but the external IP of the router is on the
same network as the * box.

Thanks!

Curt

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, November 06, 2006 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question on Aastra phones and Astrisk

Hi,
   Some odd behaviour here.  A friend and I were talking tonight,
and it seems we have both seen the same problem.   We are both using
aastra phones (I am using 9113is).We have a connection to and from
providers via SIP and IAX.When I place a call on the local hold of
the phone, and then pick them back up I can hear them, but they can
not hear me.However, if I park the call, and then pick it up
again, the audio is fine.
  Tonight I tried placing a call on hold using a Sipura/Linksys
ATA (that is just hitting 'flash', which basically puts the call on
local hold and starts music).The problem did not manifest itself.

Has anyone else had this issue?  Do you have a fix for it?  It is an
astrisk issue or an aastra issue?
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[asterisk-users] Asterisk SMS: Experience with EMS?

[Anselm Martin Hoffmeister]

Hello *,

I recently started playing around with the "SMS" application. Several of
my SIP clients are FritzBoxes, with SMS capable DECT phones connected
via ISDN or analog line.

So far, SMSing works great: I defined extension 0193010[01] to receive
SMSes, which works well with the default settings for German phones.
Incoming SMS in the morx queue will be parsed by a cronjob and forwarded
either to the recipient or to the local handler (e.g. SMS recipient 118
is the enquiries line, where phone book lookups can be got from; call
forwarding settings can be queried by texting to extension 999 etc).
It's marvellous.

But I would like to send EMS, especially bitmap files for the phones
here that in theory allow pictures and tones to be updated via SMS. I
found some EMS specs, but all that leaves me without too much clue.

Has anyone got SMS working to send bitmaps and tones to phones?

Thanks,
Anselm

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RE: [asterisk-users] Problem: 2 second silence at the beginning ofmostcalls

[Mike]

I _had_ canreinvite=yes, before I read your post.  My 
production environement though cannot handle reinvites (all phones are behind 
different NATs, too messy). So I've set those to canreinvite=no.  

 
Unfortunately, it's not making a difference.  I still 
get the 1-2 seconds silence at the beginning of my calls.  My Asterisk 
server is not behind a NAT, so in theory it should work flawlessly.  Also, 
the latency between my LAN and my Asterisk server is about 10ms, very 
stable.
 
I am trying to figure it out with Ethereal (first thing I 
did) but I'm not sure what to look for.
 
Mike

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Steve 
  LangstaffSent: November 7, 2006 8:08 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] Problem: 2 second silence at the beginning 
  ofmostcalls
  
  I was wondering whether you have canreinvite=yes on those 
  phones, and that the audio between the phones is working, but not between the 
  Asterisk server and the phones - perhaps an Ethereal trace from your Hub might 
  help?
  


From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
MikeSent: 07 November 2006 12:42To: 'Asterisk 
Users Mailing List - Non-Commercial Discussion'Subject: 
[asterisk-users] Problem: 2 second silence at the beginning of 
mostcalls

I have the 
following setup in my test lab (which reflects very much my production 
installation, just on a smaller scale)
 
Asterisk server 
- Internet -- Home router (Linksys) 
---Hub > Polycom 501 (Phone 
A)
   |---> 
Polycom 501 (Phone B)
 
 
All calls go 
through my asterisk server, even if its from one Polycom to the other. If I 
dial from phone A to phone B, audio doesnt get passed for the first 1-2 
seconds.  I end up saying "hello? hello? hello?" and eventually I heard 
something.  It makes for a bad user experience.
 
What can be the 
problem? I imagine the NAT isnt the problem, or there would be no audio at 
all.  My Asterisk is running 1.2.4, and my Polycom phones at running 
bootrom 3.2.2 and SIP 2.0.1 (fairly recent).
 
Mike
 
 
 
 
 
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Re: [asterisk-users] names of SIP aware firewalls

[Tom Rymes]

I have always been happy with Snapgear units. Most also have hardware based Encryption acceleration for IPSec and PPTP VPNs. As for setting up with SIP, use a VPN, or call their tech support line and they'll help you. Each unit comes with support for setup built-in last I checked.www.snapgear.comTom On Nov 6, 2006, at 6:24 PM, joe a. wrote:An open source firewall:www.ipcop.org lots of add ons, including a fairly new sip proxy:http://mh-lantech.css-hamburg.de/ipcop/e107_plugins/forum/forum_viewtopic.php?1847.5I think well of IPcop and the gent who did the add on, but have no experience with that particular one.joe a.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users  Tom RymesCascade Link Systemswww.cascadelinksystems.com(603) 375-1414"Intelligent technology solutions for small businesses." ___
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RE: [asterisk-users] Problem: 2 second silence at the beginning of mostcalls

[Steve Langstaff]

I was wondering whether you have canreinvite=yes on those 
phones, and that the audio between the phones is working, but not between the 
Asterisk server and the phones - perhaps an Ethereal trace from your Hub might 
help?

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  MikeSent: 07 November 2006 12:42To: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] 
  Problem: 2 second silence at the beginning of mostcalls
  
  I have the 
  following setup in my test lab (which reflects very much my production 
  installation, just on a smaller scale)
   
  Asterisk server 
  - Internet -- Home router (Linksys) ---Hub 
  > Polycom 501 (Phone A)
     |---> 
  Polycom 501 (Phone B)
   
   
  All calls go 
  through my asterisk server, even if its from one Polycom to the other. If I 
  dial from phone A to phone B, audio doesnt get passed for the first 1-2 
  seconds.  I end up saying "hello? hello? hello?" and eventually I heard 
  something.  It makes for a bad user experience.
   
  What can be the 
  problem? I imagine the NAT isnt the problem, or there would be no audio at 
  all.  My Asterisk is running 1.2.4, and my Polycom phones at running 
  bootrom 3.2.2 and SIP 2.0.1 (fairly recent).
   
  Mike
   
   
   
   
   
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RE: [asterisk-users] Queue time out

[Dean Bath]

What settings are you using when you call the queue? Ie in the
extensions.conf I have the below, this keeps calls in the queue for 30mins
(1800 secs). If you adjust it to 30, the call will come out of the queue in
30secs and move onto the next dialplan.

exten => 8000,3,Queue(fservices1800)

Regards,
Dean.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of rachid
Sent: 07 November 2006 12:19
To: asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Queue time out

Hello
   
I have set the time out to 30s in queue.conf, but
 my agent has called 2 times, and the next extension(the Hangup)
 is called after 60s.

  do you think that it is normal?

 On asterisk console i have these messages:

 -- Executing Queue("SIP/1-0cf8", "queue|tn") in new stack
 -- Started music on hold, class 'default', on SIP/1-0cf8
 -- Called SIP/adriana
 -- Nobody picked up in 3 ms
 -- Called SIP/adriana
 -- Nobody picked up in 3 ms
 -- Exiting on time-out cycle
 -- Executing Hangup("SIP/1-0cf8", "") in new stack

My config are:

queues.conf

[queue]

music=default
strategy=ringall
timeout=30
maxlen=1
context=mbdsys
announce-frequency=0
announce-holdtime=no
joinempty=strict
member => SIP/adriana,1

extension.conf
---

exten=> 999,1,Answer()
exten=> 999,2,Queue(queue|tn)
exten=> 999,3,Hangup()

thank you.


Rachid


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[asterisk-users] Problem: 2 second silence at the beginning of most calls

[Mike]

I have the following 
setup in my test lab (which reflects very much my production installation, just 
on a smaller scale)
 
Asterisk server 
- Internet -- Home router (Linksys) ---Hub 
> Polycom 501 (Phone A)
   |---> 
Polycom 501 (Phone B)
 
 
All calls go through 
my asterisk server, even if its from one Polycom to the other. If I dial from 
phone A to phone B, audio doesnt get passed for the first 1-2 seconds.  I 
end up saying "hello? hello? hello?" and eventually I heard something.  It 
makes for a bad user experience.
 
What can be the 
problem? I imagine the NAT isnt the problem, or there would be no audio at 
all.  My Asterisk is running 1.2.4, and my Polycom phones at running 
bootrom 3.2.2 and SIP 2.0.1 (fairly recent).
 
Mike
 
 
 
 
 
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Re: [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?

[R.R. Libera]

Yes, Telefonica is able to do PRI but just in a very restrictive area.

RR Libera

Ilan Rabinovitch escribió:

We briefly used it with iPlan, but found that there were some problems
with the stock asterisk implementation and Argentina variation of R2.
We ended up convincing iPlan to switch us to PRI.  As soon as we
switched to PRI all problems disappeared.

Any idea if Telefonica will be able to do PRI instead of R2?

On 10/11/06, R.R. Libera <[EMAIL PROTECTED]> wrote:

Hello,

Has somebody installed this configuration: Asterisk + E1 with MFC/R2
(Telefónica Argentina) in Argentina before? I need to know if it´s
possible with MFC/R2 argentine variation.

Thanks in advance.

R.R. Libera
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Re: [asterisk-users] Queue time out

[rachid]

Hello
  
I have set the time out to 30s in queue.conf, but

my agent has called 2 times, and the next extension(the Hangup)
is called after 60s.

 do you think that it is normal?

On asterisk console i have these messages:

-- Executing Queue("SIP/1-0cf8", "queue|tn") in new stack
-- Started music on hold, class 'default', on SIP/1-0cf8
-- Called SIP/adriana
-- Nobody picked up in 3 ms
-- Called SIP/adriana
-- Nobody picked up in 3 ms
-- Exiting on time-out cycle
-- Executing Hangup("SIP/1-0cf8", "") in new stack

My config are:

queues.conf

[queue]

music=default
strategy=ringall
timeout=30
maxlen=1
context=mbdsys
announce-frequency=0
announce-holdtime=no
joinempty=strict
member => SIP/adriana,1

extension.conf
---

exten=> 999,1,Answer()
exten=> 999,2,Queue(queue|tn)
exten=> 999,3,Hangup()

thank you.


Rachid


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[asterisk-users] Re: Pass through

[Szabó András]

Hi!Then please just tell me if it's even possible, as i cannot find any configuration to allow unknown codecs to be used in reinvited calls
.My question is that is it possible or impossible to handle this with asterisk?Thanks!AndrásOn 11/5/06, Szabó András <
[EMAIL PROTECTED]> wrote:Hi!I want to tell asterisk to
simply pass-through any codecs that my phones support. I have to use
codecs that are not popular and implemented by a third-party, asterisk
has nothing to do with them.
I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300
phones to support only g722 and asterisk declined the sip invitation.
That is bad for me. Is it possible that asterisk asks the called phone
not the codecs that asterisk supports but the codecs that the calling
phone supports?
What i want:phone1->asterisk: hello, i'm calling phone2, codecs possible: g722asterisk->phone2: hello, you have a call from phone1, codecs possible: g722phone2->asterisk: ok, let it be g722
... chit chat ...But asterisk does this:phone1->asterisk: hello, i'm calling phone2, codecs possible: g722asterisk->phone2: hello, you have a call from phone1, codecs possible: alaw, ulaw etc. (but not g722)
phone2->asterisk: no way, media not supported! (cannot agree in a codec)asterisk->phone1: beep beep beepAny ideas? Is it possible? Is it not possible?I've searched over 


voip-info.org, asterisk docs and coundn't find anything about the exact configuration.Thanks!András


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Re: [asterisk-users] Upgrading sox

[Tzafrir Cohen]

On Tue, Nov 07, 2006 at 11:14:43AM +0100, René Christensen wrote:
> Hi,
> 
> I'm currently running an * version 1.2.13 and sox version 12.17.5. I want 
> to upgrade sox to the newest release ( 12.18.2 ); need mp3 support.
> But how do I make the upgrade.
> Do I need to recompile asterisk afterwards?

No. Asterisk is not linked with sox.

> If I make a " sox -h" after a reboot  I can see the new version is running 
> but is that enough?

A reboot should not be needed either.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Mapping CLI'S in Dialplan

[Benjamin Jacob]

Your offnet calls will be more than 4 digits, so use that to ur advantage.
so, for internal calls,
exten => _,1,Set(CALLERID(all)=Name ${CALLERID(num)})
or if u dont want to change the CLID at all.. dont do anything..
exten => _,1,NoOp(nothing)

else, for all external calls(>4 digits)
exten => _X.,1,Set(CALLERID(num)=)


cheerz
- Ben.

Scott Pinhorne wrote:


Hi All

 

I am not sure what I wish to do it possible but I would like to see if 
you guys know any better.


 


I have a site who has the extensions: 1231, 1232. 1233, 1234

 

Each of these users can dial each other on the extension number an 
also has an external CLI mapped to them.


On all internal calls or calls to services such as call forwarding 
their Caller ID is: Name 


 

What I would like to have happen is have the Caller ID changed to the 
CLI only when they make an offnet call.


So what I am saying is I need to match an extension to a CLI and reset 
the Caller ID.


 


Many Thanks

SP

 

 




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[asterisk-users] Mapping CLI'S in Dialplan

[Scott Pinhorne]

Hi All

 

I am not sure what I wish to do it possible but I would like
to see if you guys know any better.

 

I have a site who has the extensions: 1231, 1232. 1233, 1234

 

Each of these users can dial each other on the extension
number an also has an external CLI mapped to them.

On all internal calls or calls to services such as call
forwarding their Caller ID is: Name 

 

What I would like to have happen is have the Caller ID
changed to the CLI only when they make an offnet call.

So what I am saying is I need to match an extension to a CLI
and reset the Caller ID.

 

Many Thanks

SP

 

 






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[asterisk-users] Upgrading sox

[René Christensen]

Hi,

I'm currently running an * version 1.2.13 and sox version 12.17.5. I want to 
upgrade sox to the newest release ( 12.18.2 ); need mp3 support.

But how do I make the upgrade.
Do I need to recompile asterisk afterwards?
If I make a " sox -h" after a reboot  I can see the new version is running 
but is that enough?


_
Log på MSN Messenger direkte på nettet  http://webmessenger.msn.com/

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Re: [asterisk-users] Dial plan Question

[Anselm Martin Hoffmeister]

Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
> I am trying to do something that I see describe in a book and it is not
> working
> 
> In my sip.conf, I have in my [fxo] context=from-pstn
> 
> I then have in extensions.conf
> 
> [from-pstn]
> 
> exten s,1,answer()
> exten s,2,playback(blah)
> 
> etc.
> 
> It never answers but if I do this
> 
> [from-pstn]
> 
> exten _x.,1,answer()
> exten _x.,2,playback(blah)
> 
> it works.  Why does the 's' extension not work here?

If "fxo" means your SIP provider, and you register with him, a specific
extension will be called. Which one shall be called can be selected by
the last parameter of the register statement, e.g.

register => 075741:[EMAIL PROTECTED]:5060/492281234567

will cause the incoming calls to appear in extension 492281234567.

Comes in handy if you have several accounts with a single SIP provider:
This way, you can simply distinguish the outward phone number for which
the call came in.

BR
Anselm

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