Re: [asterisk-users] MOH distorted voice in Native and MP3 format

[MohammedShehzad]

> Something you may want to try (its fixed it for us) is putting an I
> (uppercase I) on the asterisk invocation line.
>
> We run servers in the cloud and can't get reliable timing from ISDN
> cards etc so this instructs asterisk to generate its own internal
> timing. If you have ISDN you probably don't want to do this as they
> "should" provide better timing.
>
> Its probably worth a try anyway.
>
> eg.
> asterisk -vvvg
> change to
> asterisk -vvvgI

Something is better than nothing, I have configured the init.d script
to start asterisk with option -I, and restarted the asterisk.
Thanks Kevin for your tip.

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[asterisk-users] ringback tone after MOH, before queue member bridged

[adamk]

Good morning,

i've noticed many times that there are IVRs that play a ring tone just 
before bridging me to an agent.  My asterisk does not behave like this 
but i've always wanted to.

I'm now playing with 1.6.2.9 and i've read in queue's doc:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue

R — stops moh and rings once an agent is ringing (Asterisk Trunk)

(in queue's optinal parameters).

Could someone please explain this line to me?  I've set this option, i 
have a softphone and an ATA registered to *, pure SIP, nothing more. 
It's not working, either i'm using the r option, which disables MOH and 
just rings, or i'm using R which gives me MOH but no ringing.

It's nothing major, it just would be nice to have.

thanks
adam

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[asterisk-users] Channels not coming up

[Deepika Nijhawan]

Hi, 

 

I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi
status is not showing alarms but channels are not coming up. It is not
showing any channels when i run 'dahdi show channels'. Could anyone help
pls. 

 

 

Thanks 

Deepika

 

 

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Re: [asterisk-users] Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?

[Zhang Shukun]

Thanks. it is depends on mysqlclient.so. after i installed this module. it's ok.

2010/7/22 Gareth Blades :
> Zhang Shukun wrote:
>> hi,list
>>       Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ?  after
>> i make and make install. i cant find the .so file.
>>
>> is this mean it can't install on 64bit Cent-OS. ps: it works fine on
>> the 32 bit Cent-OS
>>
>> Thanks very much!
>>
>
> I have a live system running centos 5.5 64bit and asterisk 1.2.6 with
> the addons installed for mysql support etc...
>
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-- 
Thanks for your supporting,
have a nice day.
Sucan

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Re: [asterisk-users] Channels not coming up

[Tzafrir Cohen]

On Fri, Jul 23, 2010 at 10:32:28AM +0100, Deepika Nijhawan wrote:
> Hi, 
> 
>  
> 
> I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. 

dahdi_genconf only generates configuration. It does not apply it.

Is there any demand for a script like genzaptelconf that also applies
it? I wrote one of two of those, and it seems there are two many special
cases to handle to make them generic.

-- 
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Re: [asterisk-users] Channels not coming up

[Tzafrir Cohen]

On Fri, Jul 23, 2010 at 10:32:28AM +0100, Deepika Nijhawan wrote:
> Hi, 
> 
>  
> 
> I have connected asterisk 1.6.2.6 with an IVR and ran dahdi_genconf. Dahdi
> status is not showing alarms but channels are not coming up. It is not
> showing any channels when i run 'dahdi show channels'. Could anyone help
> pls. 

I forgot:

/etc/init.d/asterisk stop # not always required. I rather avoid it
dahdi_genconf modules
/etc/init.d/dahdi start # will likely fail. Also: restart? unload
# existing?
dahdi_genconf
# For astribanks, also:
#dahdi_genconf xpporder
/etc/init.d/dahdi start
/etc/init.d/asterisk start
> 
>  
> 
>  
> 
> Thanks 
> 
> Deepika
> 
>  
> 
>  
> 

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[asterisk-users] (no subject)

[Giusy Pagliarello]

Hi, 

I have a problem with a SIP trunk between Asterisk and central OXE Alcatel,
especially sometimes are not received inbound calls with following messages:

 

-- Executing [...@test:1] AGI("SIP/800-084250f8",
"agi://127.0.0.1/test.agi") in new stack

-- AGI Script agi://127.0.0.1/test.agi completed, returning 0

== Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN'

 

I configured the sip.conf file:



[800]

type=peer

host=172.XX.XX.XX

username=test

secret=XXX

insecure=very

context=test

disallow=all

allow=alaw

allow=ulaw

 

and the extensions.conf file:

 

exten => 375,1,AGI(agi://127.0.0.1/test.agi)

 

 

I attach to this email the sip messages receveid by Asterisk when the
problem occurs.

 

Thanks for your help.

Best regards, 

GP 

<--- SIP read from 172.25.51.1:10011 --->
INVITE sip:3...@172.24.10.188;user=phone SIP/2.0
Supported: replaces,100rel
User-Agent: ABS GW v5.1.0
P-Asserted-Identity: "ISDN_T2" 
Content-Type: application/sdp
To: 
From: "ISDN_T2" ;tag=cc01ff37a60521d35da001f98edda0ac
Contact: sip:172.25.51.1
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
CSeq: 684819861 INVITE
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c
Max-Forwards: 70
Content-Length: 314

v=0
o=OXE 1279704517 1279704517 IN IP4 172.25.51.1
s=abs
c=IN IP4 172.25.51.4
t=0 0
m=audio 32712 RTP/AVP 8 0 4 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=maxptime:30
a=rtpmap:97 telephone-event/8000

<->
--- (13 headers 17 lines) ---
Sending to 172.25.51.1 : 5060 (no NAT)
Using INVITE request as basis request - 
82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
Found peer '800'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 97
Peer audio RTP is at port 172.25.51.4:32712
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 97
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xd 
(g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.25.51.4:32712
Looking for 375 in sedoc (domain 172.24.10.188)
list_route: hop: 

<--- Transmitting (no NAT) to 172.25.51.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1
From: "ISDN_T2" ;tag=cc01ff37a60521d35da001f98edda0ac
To: 
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
CSeq: 684819861 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: 
Content-Length: 0


<>
-- Executing [...@sedoc:1] AGI("SIP/800-084250f8", 
"agi://127.0.0.1/mercury.agi") in new stack
-- AGI Script agi://127.0.0.1/mercury.agi completed, returning 0
  == Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN'
Scheduling destruction of SIP dialog 
'82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.25.51.1:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 
172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1
From: "ISDN_T2" ;tag=cc01ff37a60521d35da001f98edda0ac
To: ;tag=as3455cb36
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
CSeq: 684819861 INVITE
ser-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<>
ccsedoc*CLI>
<--- SIP read from 172.25.51.1:10011 --->
ACK sip:3...@172.24.10.188;user=phone SIP/2.0
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
From: "ISDN_T2" ;tag=cc01ff37a60521d35da001f98edda0ac
To: ;tag=as3455cb36
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c
CSeq: 684819861 ACK
Content-Length: 0


<->
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[asterisk-users] Vocera Comm Badges

[Andreas Anderson]

Hi,

has someone ever got their hands on the Comm Badges from Vocera ( 
http://www.vocera.com/ ) and knows if they use anything standard and could work 
with asterisk, or does someone know an alternative to their really small, light 
devices?

Regards,

Andreas
  
_

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[asterisk-users] 488 Not Acceptable Here

[Andy Beak]

Hi,

I'm having real difficulty in getting calls to go through with 
Asterisk.  I've managed to check that my SIP connection is made to my 
provider.  Below is an email I received from them:

snipsnipsnip
I am not certain of the reason for rejection but it has to do with the 
SDP,  it does not seem to be a codec issue, the response is as you have 
seen:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 
192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017
From: "Andy" ;tag=as5c784926
To: ;tag=SD24jn898-4C46B8A2-5688CB2-0ADE2C09
Call-ID: 32d506cd3489aa81031937f467ef6...@192.168.0.14
CSeq: 102 INVITE
Reason: Q.850 ;cause=127 ;text="Interworking, unspecified"
Content-Length: 0

There looks to be a non-standard element in your SDP that is not 
supported by any of the networks.
snipsnipsnip

Which configuration file is possibly incorrect in this scenario?

What dumps are likely to be useful to me?

Thanks,
  Andy

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Re: [asterisk-users] [AsteriskNow] Errors with cleaninstall(onmainscreen when making calls)

[Albert Scholtalbers]

Hi,

I've sent the file to Danny personally. He had made some adaptations to it
and returned it. Unfortunately not 100% successful warnings didn't disapear.
Hopefully a new release will be there soon with less cryptic warnings.

Greetings,

Albert

The file in question is probably part of Flash Operator Panel, in which
> case it is readily available in many other places on the Internet already.
>
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Re: [asterisk-users] Vocera Comm Badges

[Dean Collins]

I've seen them at trade shows, I think I remember it being proprietary.

 

What about using Dect handsets?

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
 +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andreas
Anderson
Sent: Friday, 23 July 2010 7:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Vocera Comm Badges

 

Hi,

has someone ever got their hands on the Comm Badges from Vocera (
http://www.vocera.com/ ) and knows if they use anything standard and
could work with asterisk, or does someone know an alternative to their
really small, light devices?

Regards,

Andreas



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Re: [asterisk-users] POE Splitters

[Matt]

You're using phones that draw 15Watts?!?!  Let me know what brand this is so
I can stay away from them.

On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons wrote:

> There is no such device -- it's outside of the POE spec.
>
> Class 3 devices are allowed to consume at max 15.4W. Most phones are class
> 3 devices. The math just doesn't work out. Even if you used the draft
> standard for class 4 (~30W), you could still power max 2 devices at 15W/ea.
>
> -Dave
>
> On Thu, Jul 22, 2010 at 2:46 PM, Matt  wrote:
>
>> I've got an interesting situation where I have one cable run from the feed
>> area to the service area.   I have three devices that I need to power at the
>> service area.  Is anyone aware of a device that will take the POE from the
>> cable run and then allow me to split it to two or three devices at the
>> service end?
>>
>> When I search for splitter all I get are the injectors, but I figure
>> someone has to make something I realize I'll need a power adapter with
>> enough amps to power the full load at the end.
>>
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>
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Re: [asterisk-users] POE Splitters

[Matt]

It's not necessarily this simple.  There is an approximately 50-75foot cable
run through ceilings and walls (CAT5) to the location where the phones will
be.  At the phone location there is no power.

On Thu, Jul 22, 2010 at 11:33 PM, David Backeberg wrote:

> On Thu, Jul 22, 2010 at 2:46 PM, Matt  wrote:
> > I've got an interesting situation where I have one cable run from the
> feed
> > area to the service area.   I have three devices that I need to power at
> the
> > service area.  Is anyone aware of a device that will take the POE from
> the
> > cable run and then allow me to split it to two or three devices at the
> > service end?
>
> The obvious answer is "don't do that".
>
> *buy DC power bricks for the phones / devices
> *buy a small PoE switch for the area, plugged into the single ethernet
> cable as a trunk
> *pull more cable from the original endpoint
>
> Any of those three will be more reliable and predictable when
> debugging than inventing your own PoE solution. I've tried to invent
> my own PoE solution using a soldering iron and bulk ethernet cable.
> Take it from me, don't go down that road. Yes, you will learn all
> manner of interesting things about DC voltage loss over distance,
> blah, blah, blah.
>
> Your time is almost undoubtedly worth more money than you'll save by
> pursuing the 'conventional approaches'. Just don't do it.
>
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Re: [asterisk-users] POE Splitters

[Kevin P. Fleming]

On 07/23/2010 02:46 PM, Matt wrote:
> It's not necessarily this simple.  There is an approximately 50-75foot
> cable run through ceilings and walls (CAT5) to the location where the
> phones will be.  At the phone location there is no power.

I thought it was fairly obvious, but search for "PoE extractor". Here's
an example:

http://www.shireeninc.com/poe-extractor.html

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] POE Splitters

[David Backeberg]

On Fri, Jul 23, 2010 at 8:46 AM, Matt  wrote:
> It's not necessarily this simple.  There is an approximately 50-75foot cable
> run through ceilings and walls (CAT5) to the location where the phones will
> be.  At the phone location there is no power.

You always have options. You just have to decide what is more difficult:

* moving the phone/devices somewhere else. Easiest solution.
* having an electrician pull AC power to the location, then use DC
power bricks or PoE switch
* having a data cable person pull more ethernet to the location

If you already have one ethernet cable that managed to make that 50-75
foot run, then clearly it can be done, and a professional could even
use that cable to yank three more along the same run, and then you're
all set.

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Re: [asterisk-users] [AsteriskNow] Errors with cleaninstall(onmainscreen when making calls)

[Danny Nicholas]

Sorry my changes didn't solve the problem;  for reference, this is what I
did:
3360,3367c3360
< my $key;
< if ( "$server^$hash_temporal{'Destination'}") {
<$key = "$server^$hash_temporal{'Destination'}";
<}
< else {
<$key="0";
<}
<
---
> my $key  = "$server^$hash_temporal{'Destination'}";
3374,3377c3367,3368
< if ( $chanvar{ $hash_temporal{"SrcUniqueID"} }{$var} ) {
 if ( defined( $chanvar{ $hash_temporal{"SrcUniqueID"} }{$var}
) ) {
> $passvar{ $hash_temporal{"DestUniqueID"} }{$var} =
$chanvar{ $hash_temporal{"SrcUniqueID"} }{$var};
3383,3386d3373
<  if ( ! $hash_temporal{'Source'} =~ m/^Local/i ) {
< $hash_temporal{'Source'} = 'Remote';
< }
Since the messages were coming from missing elements of $hash_temporal, this
seemed like a reasonable approach to the problem.


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Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes

[Steve Underwood]

On 07/23/2010 11:17 AM, Alexander Aksarin wrote:
> On 21:46 Thu 22 Jul , Steve Underwood wrote:
>
>> It might help if you explained what you expect those pages should look
>> like. I see three quite plausible pages.
>>  
> I expect to see this http://imagebin.ca/img/Eihpy0.jpg
>
>
That's just how your images look for me, so I guess your problem is 
described here http://www.soft-switch.org/spandsp_faq/ar01s09.html

Steve


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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[Paul Belanger]

On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce  wrote:
> Any help is appreciated.
>
Are you explicitly calling Hangup() within your dial-plans?

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Re: [asterisk-users] [AsteriskNow] Errors with cleaninstall(onmainscreen when making calls)

[Albert Scholtalbers]

Long live diff. Yup it seem it would.
Would there be a temp solution to route all errors to a file in
/var/log/op_server or so
In that why the screen stays clean until the source of the warnings/errors
is found.
We think it has to do with a setting in the config file, maybe all options
have to be enabled to make it work

Albert

3360,3367c3360
> < my $key;
> < if ( "$server^$hash_temporal{'Destination'}") {
> <$key = "$server^$hash_temporal{'Destination'}";
> <}
> < else {
> <$key="0";
> <}
> <
> ---
> > my $key  = "$server^$hash_temporal{'Destination'}";
> 3374,3377c3367,3368
> < if ( $chanvar{ $hash_temporal{"SrcUniqueID"} }{$var} ) {
>  }{$var} ) ) {
> <$passvar{ $hash_temporal{"DestUniqueID"} }{$var} =
> $chanvar{ $hash_temporal{"SrcUniqueID"} }{$var};
> <}
> ---
> > if ( defined( $chanvar{ $hash_temporal{"SrcUniqueID"} }{$var}
> ) ) {
> > $passvar{ $hash_temporal{"DestUniqueID"} }{$var} =
> $chanvar{ $hash_temporal{"SrcUniqueID"} }{$var};
> 3383,3386d3373
> <  if ( ! $hash_temporal{'Source'} =~ m/^Local/i ) {
> < $hash_temporal{'Source'} = 'Remote';
> < }
> Since the messages were coming from missing elements of $hash_temporal,
> this
> seemed like a reasonable approach to the problem.
>
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] [AsteriskNow] Errors withcleaninstall(onmainscreen when making calls)

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Scholtalbers
Sent: Friday, July 23, 2010 8:27 AM



 

Long live diff. Yup it seem it would.
Would there be a temp solution to route all errors to a file in
/var/log/op_server or so
In that why the screen stays clean until the source of the warnings/errors
is found.
We think it has to do with a setting in the config file, maybe all options
have to be enabled to make it work

Albert

Long shot, but is loguniqueid=yes in your cdr.conf?

 

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[asterisk-users] calls don't hang up correctly on VM

[Danny Nicholas]

Hello List,

   I'm moving my asterisk testing installation from CENTOS 5.4
on a real machine to SUSE on a xen VM.  Everything seemed to go off without
a hitch until I really looked at it.  The call answers and processes
correctly, but when it is time to end the call,  the phone never disconnects
from asterisk.  For a simple functionality test, I use this "Monkeys"
snippet to tell me if all is well:

dialplan show 9...@default

[ Context 'default' created by 'pbx_config' ]

  '99' =>   1. Answer()
[pbx_config]

2. Playback(tt-monkeys)
[pbx_config]

3. Noop(tt-monkeys)
[pbx_config]

4. Hangup()
[pbx_config]

 

-= 1 extension (4 priorities) in 1 context. =-

This is my CLI output from a test call (1.6.2.9)

*CLI>   == Using SIP RTP CoS mark 5

-- Executing [...@default:1] Answer("SIP/170-", "") in new stack

-- Executing [...@default:2] Playback("SIP/170-", "tt-monkeys")
in new stack

--  Playing 'tt-monkeys.gsm' (language 'en')

-- Executing [...@default:3] NoOp("SIP/170-", "tt-monkeys") in
new stack

-- Executing [...@default:4] Hangup("SIP/170-", "") in new stack

  == Spawn extension (default, 99, 4) exited non-zero on 'SIP/170-'

-- Executing [...@default:1] Set("SIP/170-", "CDR(userfield)=
Hangupcause:16") in new stack

-- Executing [...@default:2] Verbose("SIP/170-", "details - time
time2  status ") in new stack

details - time  time2  status

-- Executing [...@default:3] GotoIf("SIP/170-", "0?end-call,s,1")
in new stack

-- Executing [...@default:4] Verbose("SIP/170-", "details - time
time2  status ") in new stack

details - time  time2  status

-- Executing [...@default:5] NoOp("SIP/170-", "id 1279891796.0
time 16") in new stack

-- Executing [...@default:6] NoOp("SIP/170-", "caller hung up eh")
in new stack

-- Executing [...@default:7] Goto("SIP/170-", "end-call,s,1") in
new stack

-- Goto (end-call,s,1)

-- Executing [...@end-call:1] NoOp("SIP/170-", "Verbose(details -
time  time2  status )") in new stack

-- Executing [...@end-call:2] Hangup("SIP/170-", "") in new stack

  == Spawn extension (end-call, s, 2) exited non-zero on 'SIP/170-'

The call executed as expected, but the telephone never hangs up

If I do an IAX call, the hangup occurs as expected.

 

Any clues?

Thanks

Danny Nicholas

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[asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

[Benoit]

Hi,

For some reason (outbound call tracking) I've got a few different 
outbound call process (using a macro for queuemetrics logging, or direct 
call)
i wanted to factorise the routing process so i came up with something 
like the following. All in one it's working like expected, however
every "ael reload" command trigger a lot of warning like that

 "application call to Gosub affects flow of control, and needs to be 
re-written using AEL if, while, goto, etc. keywords instead!"

But i fail to see how i could do it another way, any idea/suggestion ?


Extraction of the outbound processing structure:

context outboundSimple {
 _9X. => {
 // prepare callerid, secret, ...
 /// ...

 // start call routing
 Gosub(pstnRouting,${EXTEN:1},1);
 // back, handle return status
&dialstatus(${DIALSTATUS},${EXTEN:1});
 }
}

context outboundQueue1 {
 _9X. => {
 // prepare callerid, secret, ...
 /// ...

&qmoutqdial(${EXTEN:1},DAHDI/g1/${EXTEN:1},queue-out,Agent/${AgentNum});
 }
}

context outboundQueue2 {

}

// outbound call logging for queuemetrics:
macro qmoutqdial( clid, dialstring, queue, agent )
{
 start_dial_time = ${EPOCH};
 QueueLog(${queue},${UNIQUEID},${agent},CALLOUTBOUND,-|${clid});

 
Set(dialopts=gWKU(queuelog_connect_event^${queue}^${UNIQUEID}^${agent}^${start_dial_time}));
 Gosub(pstnRouting,${clid},1);

 end_dial_time = ${EPOCH};
 verb = COMPLETECALLER;
&queuelog_hangup_event(${queue},${UNIQUEID},${agent},${start_dial_time});
 return;
}


// central call routing rules
context pstnRouting {

 _06 => { Gosub(pstnInterface2,${EXTEN},1); Return; }
 

 // left over
 _X. =>  { Gosub(pstnInterface1,${EXTEN},1); Return; }
}

context pstnInterface1 {
 _X. => {
 // setup the interface callerid/secret status

 ChanIsAvail(DAHDI/g1);
 if( ! ${ISNULL(${AVAILORIGCHAN})} ) {
 Dial(DAHDI/g1/${ext},,${dialopts});
 }
 Return;
 }
}

context pstnInterface2 {
 _X. => {
 // setup the interface callerid/secret status

 ChanIsAvail(SIP/);
 if( ! ${ISNULL(${AVAILORIGCHAN})} ) {
 Dial(SIP/.../${ext},,${dialopts});
 }
 Return;
 }
}


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Re: [asterisk-users] [AsteriskNow] Errors withcleaninstall(onmainscreen when making calls)

[Albert Scholtalbers]

No value was set in  /etc/asterisk/cdr.conf added it, but without succes

Long shot, but is loguniqueid=yes in your cdr.conf?
>
>
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[asterisk-users] Poor-man's paging through multiple phones?

[Peter Pauly]

We're mostly Cisco CallManager with some SIP and Asterisk.

I want someone at one of our locations to be able to dial and number
and have Asterisk simultaneously dial several Call-Manager extensions
which are set to auto-answer and talk into the phone creating a sort
of paging system.

We have informacast, but it is too cumbersome for the users.

I know Asterisk can ring several phones at the same time... if one of
them answers, the others stop right?

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Re: [asterisk-users] Vocera Comm Badges

[Philipp von Klitzing]

Hi!

> I´ve seen them at trade shows, I think I remember it being proprietary.
> What about using Dect handsets?

That Star Trek device has always interested me. Too bad they chose WiFi 
over DECT, though.

Vocera badge:
* WLAN b/g
* Talktime 2-2.5 hours, standby 20-27 hours
* headset jack
* OLED display (why don't they ever show this on the pics)
* Linux based

Back side:
http://farm4.static.flickr.com/3412/3496101115_32319840a5.jpg

System:
* Windows server
* Nuance ASR & biometrics & Dictaphone
* Dialogic T1/ISDN/Analog cards
* SIP interface
* iPhone app

I am also curious to hear some user reports, in particular about battery 
performance: Does it last one entire working day, or does it need to be 
exchanged (once?) per day? What is the smallest feasible installation? 
What speech protocol does the badge use to talk to the server?

"First quarter of 2010 was the third consecutive profitable quarter for 
the company. In addition, since the beginning of 2010, the company has 
announced the addition of 22 new employees across North America and the 
United Kingdom."

Philipp


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Re: [asterisk-users] ringback tone after MOH, before queue member bridged

[Jason Aarons (US)]

I normally work with other 3rd party IVRs, usually once the Agent is Reserved 
we signal the phone system to play Music on Hold while it's ringing the Agent.  
The trick here is to replace the Music on Hold with a fake ring file.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ad...@3a.hu
Sent: Friday, July 23, 2010 3:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ringback tone after MOH, before queue member bridged

Good morning,

i've noticed many times that there are IVRs that play a ring tone just before 
bridging me to an agent.  My asterisk does not behave like this but i've always 
wanted to.

I'm now playing with 1.6.2.9 and i've read in queue's doc:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue

R — stops moh and rings once an agent is ringing (Asterisk Trunk)

(in queue's optinal parameters).

Could someone please explain this line to me?  I've set this option, i have a 
softphone and an ATA registered to *, pure SIP, nothing more. 
It's not working, either i'm using the r option, which disables MOH and just 
rings, or i'm using R which gives me MOH but no ringing.

It's nothing major, it just would be nice to have.

thanks
adam

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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[bruce bruce]

This is running Elastix (FreePBX), so I am pretty sure there is Hangup()
requested. At least this doesn't happen ALL THE TIME. So, something is
getting stuck.

Thanks,
Bruce

On Fri, Jul 23, 2010 at 9:10 AM, Paul Belanger  wrote:

> On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce  wrote:
> > Any help is appreciated.
> >
> Are you explicitly calling Hangup() within your dial-plans?
>
> --
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> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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Re: [asterisk-users] POE Splitters

[bruce bruce]

You can also use Ethernet Over Power Lines solution or wireless :-)

On Fri, Jul 23, 2010 at 8:55 AM, David Backeberg wrote:

> On Fri, Jul 23, 2010 at 8:46 AM, Matt  wrote:
> > It's not necessarily this simple.  There is an approximately 50-75foot
> cable
> > run through ceilings and walls (CAT5) to the location where the phones
> will
> > be.  At the phone location there is no power.
>
> You always have options. You just have to decide what is more difficult:
>
> * moving the phone/devices somewhere else. Easiest solution.
> * having an electrician pull AC power to the location, then use DC
> power bricks or PoE switch
> * having a data cable person pull more ethernet to the location
>
> If you already have one ethernet cable that managed to make that 50-75
> foot run, then clearly it can be done, and a professional could even
> use that cable to yank three more along the same run, and then you're
> all set.
>
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Re: [asterisk-users] Poor-man's paging through multiple phones?

[Danny Nicholas]

>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>Subject: [asterisk-users] Poor-man's paging through multiple phones?

>We have informacast, but it is too cumbersome for the users.

>I know Asterisk can ring several phones at the same time... if one of
>them answers, the others stop right?

Dial stops when the first line answers or timeout occurs, so
Dial(dahdi/1/5551212&dahdi/2/5550123&dahdi/3/5550231,60,m) would stop after
60 seconds or as soon as anybody picked up.



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Re: [asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

[Zeeshan Zakaria]

Hi,

I try to avoid any warnings, as they can  turn into errors later.

I remember having problems with GoSub long time ago, don't remember what it
was, but I moved to macros after that.

For what you are trying to achieve, I use macros. Just jump to a macro,
evaluate what you need to, save the results in variables, and use these
variables in the calling context to proceed further. For example, in my
context for outbound calls, called [outbound], I jump to macro
[user-account-info], get account info, come back to [outbound], then jump to
[blacklist], check the caller ID against a blacklist, come back to
[outbound], jump to two other macros, and finally I have all the info I need
to proceed with a call, stored in various variables. Based on these
variables I trigger the Dial command, or do something else as needed.

In your case, you can make it even simpler, e.g. like this:

context outboundSimple {
_9X. => {
// prepare callerid, secret, ...
/// ...

// start call routing
if("${EXTEN:1:3}"="06") {
&pstnInterface2(${EXTEN:1});
}
else {
&pstnInterface1(${EXTEN:1});
};
// back, handle return status
};

h => {
&dialstatus(${DIALSTATUS},${EXTEN:1});
};

}


macro pstnInterface1(number) {

// setup the interface callerid/secret status

ChanIsAvail(DAHDI/g1);
if( ! ${ISNULL(${AVAILORIGCHAN})} ) {
Dial(DAHDI/g1/${number},,${dialopts});
};

catch h {
&dialstatus(${DIALSTATUS},${number});
};

}

macro pstnInterface2(number) {

// setup the interface callerid/secret status

ChanIsAvail(SIP/);
if( ! ${ISNULL(${AVAILORIGCHAN})} ) {
Dial(SIP/.../${number},,${dialopts});
};

catch h {
&dialstatus(${DIALSTATUS},${number});
};
}

On Fri, Jul 23, 2010 at 9:50 AM, Benoit  wrote:

>
> Hi,
>
> For some reason (outbound call tracking) I've got a few different
> outbound call process (using a macro for queuemetrics logging, or direct
> call)
> i wanted to factorise the routing process so i came up with something
> like the following. All in one it's working like expected, however
> every "ael reload" command trigger a lot of warning like that
>
> "application call to Gosub affects flow of control, and needs to be
> re-written using AEL if, while, goto, etc. keywords instead!"
>
> But i fail to see how i could do it another way, any idea/suggestion ?
>
>
> Extraction of the outbound processing structure:
>
> context outboundSimple {
> _9X. => {
> // prepare callerid, secret, ...
> /// ...
>
> // start call routing
> Gosub(pstnRouting,${EXTEN:1},1);
> // back, handle return status
> &dialstatus(${DIALSTATUS},${EXTEN:1});
> }
> }
>
> context outboundQueue1 {
> _9X. => {
> // prepare callerid, secret, ...
> /// ...
>
> &qmoutqdial(${EXTEN:1},DAHDI/g1/${EXTEN:1},queue-out,Agent/${AgentNum});
> }
> }
>
> context outboundQueue2 {
> 
> }
>
> // outbound call logging for queuemetrics:
> macro qmoutqdial( clid, dialstring, queue, agent )
> {
> start_dial_time = ${EPOCH};
> QueueLog(${queue},${UNIQUEID},${agent},CALLOUTBOUND,-|${clid});
>
>
>
> Set(dialopts=gWKU(queuelog_connect_event^${queue}^${UNIQUEID}^${agent}^${start_dial_time}));
> Gosub(pstnRouting,${clid},1);
>
> end_dial_time = ${EPOCH};
> verb = COMPLETECALLER;
> &queuelog_hangup_event(${queue},${UNIQUEID},${agent},${start_dial_time});
> return;
> }
>
>
> // central call routing rules
> context pstnRouting {
>
> _06 => { Gosub(pstnInterface2,${EXTEN},1); Return; }
> 
>
> // left over
> _X. =>  { Gosub(pstnInterface1,${EXTEN},1); Return; }
> }
>
> context pstnInterface1 {
> _X. => {
> // setup the interface callerid/secret status
>
> ChanIsAvail(DAHDI/g1);
> if( ! ${ISNULL(${AVAILORIGCHAN})} ) {
> Dial(DAHDI/g1/${ext},,${dialopts});
> }
> Return;
> }
> }
>
> context pstnInterface2 {
> _X. => {
> // setup the interface callerid/secret status
>
> ChanIsAvail(SIP/);
> if( ! ${ISNULL(${AVAILORIGCHAN})} ) {
> Dial(SIP/.../${ext},,${dialopts});
> }
> Return;
> }
> }
>
>
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Re: [asterisk-users] POE Splitters

[Kevin P. Fleming]

On 07/23/2010 04:40 PM, bruce bruce wrote:
> You can also use Ethernet Over Power Lines solution or wireless :-)

His issue wasn't getting the network connection delivered, it was the
power :-)

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Dahdi dial plan

[Sebastian Schwardt]

Hi,

 

can anybody please show me a valid dial plan for a dahdi card with a bri
port? I can not get asterisk 1.6 to dial a number. I want to receive one
call on the first b channel and dial another number on the second b channel
of the same isdn port.

 

I tried something like Dial(DAHDI/g1/123456789) but that does not work. It
always says that the channel is busy.

 

Kind regards,

 

Sebastian Schwardt

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Re: [asterisk-users] POE Splitters

[Gordon Henderson]

On Fri, 23 Jul 2010, Matt wrote:

> It's not necessarily this simple.  There is an approximately 50-75foot cable
> run through ceilings and walls (CAT5) to the location where the phones will
> be.  At the phone location there is no power.

Why not use analogue phones? Get some nice ones with caller ID display and 
use a multi-port ATA. I did that for a site recently - they had power, but 
the cable run was 175m. One cat-5 = 4 pairs = 4 phones. How many do you 
need?

Gordon

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Re: [asterisk-users] Poor-man's paging through multiple phones?

[Zeeshan Zakaria]

There is a 'Page' command in asterisk for this purpose.

What you are trying to achieve, I have implemented a few times using MeetMe.
But I needed to send a sip-info message to customers' grandstream phones to
turn speakers on their speaker on. Do you have some similar option on your
phones?

Zeeshan A Zakaria

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On 2010-07-23 10:49 AM, "Danny Nicholas"  wrote:

>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>Subject: [asterisk-users] Poor-man's paging through multiple phones?


>We have informacast, but it is too cumbersome for the users.

>I know Asterisk can ring several ph...
Dial stops when the first line answers or timeout occurs, so
Dial(dahdi/1/5551212&dahdi/2/5550123&dahdi/3/5550231,60,m) would stop after
60 seconds or as soon as anybody picked up.




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Re: [asterisk-users] Dahdi dial plan

[Tzafrir Cohen]

On Fri, Jul 23, 2010 at 04:55:57PM +0200, Sebastian Schwardt wrote:
> Hi,
> 
>  
> 
> can anybody please show me a valid dial plan for a dahdi card with a bri
> port? I can not get asterisk 1.6 to dial a number. I want to receive one
> call on the first b channel and dial another number on the second b channel
> of the same isdn port.
> 
>  
> 
> I tried something like Dial(DAHDI/g1/123456789) but that does not work. It
> always says that the channel is busy.

Let's assume you have the folloiwng in chan_dahdi.conf:

[channels]
signalling = bri_cpe_ptmp
switchtype = euroisdn
context = from-bri
group = 1
channel => 1-2


Now, in extensions.conf:

[from-bri]; Handle inclming calls from ISDN
; Example of special handling for a specific MSN:
exten => 1,1122334455,NoOp(Call to MSN ${EXTEN})
exten => n,1122334455,...  Continue handling it. Answer? 

; A catch-all extension for all of them:
exten => 1,_X.,NoOp(Call to MSN ${EXTEN})
exten => n,_X.,...  Continue handling it. Answer? 

See also the [demo] context in the sample extension.conf .



Now, suppose you have a SIP phone that is sent to context 'interlanl'
(or another context that contains it). Suppose you want to make
any number that begins with 1 and has exactly 10 digits go through the
ISDN:

[internal]
exten => _1X,Dial(DAHDI/G1/${EXTEN})


'G' means that it will first try the second BRI line and only then try
the first line. In practice I suppose it doesn't matter as if on is busy
in a call, it will use the other.

-- 
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icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Dahdi dial plan

[Danny Nicholas]

>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Schwardt
>Subject: [asterisk-users] Dahdi dial plan

 

>can anybody please show me a valid dial plan for a dahdi card with a bri
port? I can not get asterisk 1.6 to dial a number. I want to receive one
call on the first b >channel and dial another number on the second b channel
of the same isdn port.

 

>I tried something like Dial(DAHDI/g1/123456789) but that does not work. It
always says that the channel is busy.

 

>Sebastian Schwardt

 

As I recall, g1 will pick out the first line in a group and that's the line
you are on;  G1 goes in reverse, and therefore should get the first
available line in the group for dialout.

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Re: [asterisk-users] POE Splitters

[Hans Witvliet]

On Thu, 2010-07-22 at 17:41 -0500, Karl Fife wrote:

> >> enough amps to power the full load at the end.
> >>
> 
> You could do someting with passive POE--in other words not 802.2af POE, but 
> rather the 'dumb' kind of POE which just injects power on the unused pairs. 
> Passive POE (being passive) does not have a hard wattage limit per drop 
> limitation imposed by 802.2af.  On the far end, you could split out the 
> power to run a 4-port 802.11af POE switch.
> 
> Passive POE would preclude GigE, but at least you wouldn't have to add 
> ethernet drops.  In theory you could preserve GigE by looking for a IEEE 
> 802.3at [sic] switch.  IEEE 802.3at allows 36 watts per port, but good luck 
> finding (or affording) a 802.3at-powered 3-port 802.2af POE switch even if 
> all 3 downstream devices don't draw the maximum wattage simultaneously :-)
> 
afaicr there are not a lot of hard phones doing 1Gb,

And POE-switches that can do 1Gb resembles a jumbo-jet compares to other
planes; Both in price an amount of noise...

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Re: [asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

[Benoit]

  Le 23/07/2010 16:44, Zeeshan Zakaria a écrit :
> Hi,
>
> I try to avoid any warnings, as they can  turn into errors later.
well, that's exactly the point of this inquiry :)
>
> I remember having problems with GoSub long time ago, don't remember 
> what it was, but I moved to macros after that.
>
> For what you are trying to achieve, I use macros. Just jump to a 
> macro, evaluate what you need to, save the results in variables, and 
> use these variables in the calling context to proceed further. For 
> example, in my context for outbound calls, called [outbound], I jump 
> to macro [user-account-info], get account info, come back to 
> [outbound], then jump to [blacklist], check the caller ID against a 
> blacklist, come back to [outbound], jump to two other macros, and 
> finally I have all the info I need to proceed with a call, stored in 
> various variables. Based on these variables I trigger the Dial 
> command, or do something else as needed.
>
> In your case, you can make it even simpler, e.g. like this:
Already though of this, doesn't really suit the need.

The extract was only a really short version of the original system, the 
routing is a bit more complex than that
and redoing it using if/switch won't be very efficient, especially since 
context are really made for this.


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Re: [asterisk-users] application call to Gosub affects flow of control, and needs to be re-written using AEL

[Zeeshan Zakaria]

Then I would suggest using the method I mentioned earlier, i.e. using
macros. I have a really sophisticated dialplan for my multi-tenant system,
which also incorporates some serious security stuff, along with call
routing, trunk selection decisions and other checks, and for me macros work
really well.

Zeeshan A Zakaria

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On 2010-07-23 11:40 AM, "Benoit"  wrote:

 Le 23/07/2010 16:44, Zeeshan Zakaria a écrit :

> Hi,
>
> I try to avoid any warnings, as they can turn into errors later.
well, that's exactly the point of this inquiry :)

>
> I remember having problems with GoSub long time ago, don't remember
> what it was, but I moved ...
Already though of this, doesn't really suit the need.

The extract was only a really short version of the original system, the
routing is a bit more complex than that
and redoing it using if/switch won't be very efficient, especially since
context are really made for this.


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Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

[AMARDEEP SINGH]

Our SMS-gateway is not PSTN accessible.

On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese  wrote:

>  AMARDEEP SINGH wrote:
>
> Hello All,
>
> Scenario:
> -We use asterisk as voicemail server for our cellular network. Asterisk box
> is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip.
> -Extensions in * are virtual, just for leaving and accessing voicemail.
>
> Requirement:
> Asterisk to send SMS to cell switch(running SMSC) on reception of new
> voicemail.
>
> Pointers required from Maillist users:
> -How can I make * talk to SMSC(ip address:port).
> -Anyone using similar topology?
> -there are not enough examples/man/maillist of using app_sms(), smsq.
>
> Thanks:
> -A
>
> qpage?
>
>
>
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[asterisk-users] Asterisk 1.6.2.10 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 1.6.2.10.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.2.10 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are a few of the issues resolved by community developers:

  * Allow users to specify a port for DUNDI peers.
(Closes issue #17056. Reported, patched by klaus3000)

  * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER 
is
set.
(Closes issue #16815. Reported, patched by rain)

  * If there is realtime configuration, it does not get re-read on reload unless
the config file also changes.
(Closes issue #16982. Reported, patched by dmitri)

  * Send AgentComplete manager event for attended transfers.
(Closes issue #16819. Reported, patched by elbriga)

  * Correct manager variable 'EventList' case.
(Closes issue #17520. Reported, patched by kobaz)

In addition, changes to res_timing_pthread that should make it more stable have
also been implemented.

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10

Thank you for your continued support of Asterisk!

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[asterisk-users] Asterisk 1.4.34 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 1.4.34.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.34 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are a few of the issues resolved by community developers:

  * Allow users to specify a port for DUNDi peers.
(Closes issue #17056. Reported, patched by klaus3000)

  * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER 
is
set.
(Closes issue #16815. Reported, patched by rain)

  * First caller into a dynamic conference new enters the pin once.
(Closes issue #15878. Reported, patched by pabelanger)

  * Send AgentComplete manager events in the event of blind and attended
transfers.
(Closes issue #16819. Reported, patched by elbriga)

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] 488 Not Acceptable Here

[Jamie A. Stapleton]

A packet capture would be most useful.  Then, you could review your SDP with 
your provider.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Beak
Sent: Friday, July 23, 2010 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 488 Not Acceptable Here

Hi,

I'm having real difficulty in getting calls to go through with 
Asterisk.  I've managed to check that my SIP connection is made to my 
provider.  Below is an email I received from them:

snipsnipsnip
I am not certain of the reason for rejection but it has to do with the 
SDP,  it does not seem to be a codec issue, the response is as you have 
seen:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 
192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017
From: "Andy" ;tag=as5c784926
To: ;tag=SD24jn898-4C46B8A2-5688CB2-0ADE2C09
Call-ID: 32d506cd3489aa81031937f467ef6...@192.168.0.14
CSeq: 102 INVITE
Reason: Q.850 ;cause=127 ;text="Interworking, unspecified"
Content-Length: 0

There looks to be a non-standard element in your SDP that is not 
supported by any of the networks.
snipsnipsnip

Which configuration file is possibly incorrect in this scenario?

What dumps are likely to be useful to me?

Thanks,
  Andy

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Re: [asterisk-users] POE Splitters

[Philip Prindeville]

 Sounds like a great ear warmer!!!

Hell, you can probably grill a panini with it if you're patient.

On 7/23/10 6:39 AM, Matt wrote:

You're using phones that draw 15Watts?!?!  Let me know what brand this is so I 
can stay away from them.

On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons mailto:d...@videon-central.com>> wrote:

There is no such device -- it's outside of the POE spec.

Class 3 devices are allowed to consume at max 15.4W. Most phones are class 
3 devices. The math just doesn't work out. Even if you used the draft standard 
for class 4 (~30W), you could still power max 2 devices at 15W/ea.

-Dave

On Thu, Jul 22, 2010 at 2:46 PM, Matt mailto:mhop...@gmail.com>> wrote:

I've got an interesting situation where I have one cable run from the 
feed area to the service area.   I have three devices that I need to power at 
the service area.  Is anyone aware of a device that will take the POE from the 
cable run and then allow me to split it to two or three devices at the service 
end?

When I search for splitter all I get are the injectors, but I figure 
someone has to make something I realize I'll need a power adapter with 
enough amps to power the full load at the end.



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Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

[Lyle Giese]

Maybe you need to read the man page for qpage.  The qpage client can
send the page to an SNPP server over TCP/IP.

Lyle

AMARDEEP SINGH wrote:
> Our SMS-gateway is not PSTN accessible.
>
> On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese  > wrote:
>
> AMARDEEP SINGH wrote:
>> Hello All,
>>
>> Scenario:
>> -We use asterisk as voicemail server for our cellular network.
>> Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway)
>> through sip.
>> -Extensions in * are virtual, just for leaving and accessing
>> voicemail.
>>
>> Requirement:
>> Asterisk to send SMS to cell switch(running SMSC) on reception of
>> new voicemail.
>>
>> Pointers required from Maillist users:
>> -How can I make * talk to SMSC(ip address:port).
>> -Anyone using similar topology?
>> -there are not enough examples/man/maillist of using app_sms(), smsq.
>>
>> Thanks:
>> -A
> qpage?
>
>
>
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>


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[asterisk-users] Attended Transfer question

[Warren Selby]

I've been asked to implement the following transfer workflow in an asterisk
system, and I'm not seeing an easy way to do the bolded steps below (steps 4
and 5 for those with a text-only email client):

1 - Put the call on hold
2 - Call the extension for the staff member needed
3 - Give them a rundown of the  caller and situation
*4 - Bring the caller on with the staff member the call will be transferred
to
5 - The person transferring the call will recap the situation with the
caller and staff member the call will be transferred to*
6 - Transfer the call and drop off without the call being dropped

Now, the way things work now:

1 - Press the transfer button on the phone, putting the original call on
hold.
2 - Dial the staff member needed.
3 - Explain situation.
4 - Press transfer button on phone again, transferring the caller to the
staff member, removing yourself from the call entirely.

Any suggestions?

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Attended Transfer question

[Danny Nicholas]

>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
>Subject: [asterisk-users] Attended Transfer question

 

>I've been asked to implement the following transfer workflow in an asterisk
system, and I'm not seeing an easy way to do the bolded steps below (steps 4
and 5 for those with a text-only email client):

You could create a dynamic meetme room for the 3 legs and drop out when
done.  Or do it with X static meetme rooms.  You could set up a context to
create the MM room and call the supervisor, connecting them to the room.

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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[Maurizio Faccio adinet]

I guess same trouble with Elastix 1.5.2-2.3
dahdi 2.1.0.4      19
Asterisk 1.4.25.1
Digium TDM 2400



[Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: master: 14, slave: 1,
nothingok: 0
[Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: Stopping tones on 14/0
talking to 1/0
[Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: Stopping tones on 1/0
talking to 14/0
[Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: Making 1 slave to master
14 at 0
[Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: Added 11 to conference 9/14
[Jul 23 16:47:22] DEBUG[18890] chan_dahdi.c: Added 25 to conference 9/1
[Jul 23 16:47:22] VERBOSE[18890] logger.c: -- Native bridging
DAHDI/14-1 and DAHDI/1-1

Hope it helps 

Maurizio FAccio



El 23/07/2010 11:39 a.m., bruce bruce escribió:
This is running Elastix (FreePBX), so I am pretty sure
there is Hangup() requested. At least this doesn't happen ALL THE TIME.
So, something is getting stuck.
  
  
  Thanks,
  Bruce
  
  On Fri, Jul 23, 2010 at 9:10 AM, Paul
Belanger 
wrote:
  On
Fri, Jul 23, 2010 at 1:16 AM, bruce bruce  wrote:
> Any help is appreciated.
>
Are you explicitly calling Hangup() within your dial-plans?

--
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| IRC: pabelanger (Freenode)
blog.polybeacon.com


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Re: [asterisk-users] Poor-man's paging through multiple phones?

[Elliot Otchet]

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Pauly
Sent: Friday, July 23, 2010 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Poor-man's paging through multiple phones?

We're mostly Cisco CallManager with some SIP and Asterisk.

I want someone at one of our locations to be able to dial and number and have 
Asterisk simultaneously dial several Call-Manager extensions which are set to 
auto-answer and talk into the phone creating a sort of paging system.

We have informacast, but it is too cumbersome for the users.

I know Asterisk can ring several phones at the same time... if one of them 
answers, the others stop right?

--
True, but you could have your paging app record the announcement and then play 
the recording out via an AGI/AMI Originate to each phone.

If you're looking more for a hoot-and-holler system, use the AGI/AMI Originate 
to dial the extensions and place the phones into a MeetMe/Conference bridge.

For the simultaneous dialing part, be sure to research the Async option on 
Originate.  The increased performance is quite noticeable in applications like 
this.

PERL/C/PHP/'Rails/Java, choose your weapon.  If you've got the CM set up 
already, each of the AGI/AMI frameworks should have examples on how to do 
Originate calls.

-Elliot





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[asterisk-users] voicemail

[mattias]

Can i add functions to voicemail
Like some companies have  the ability to press 6 to call the people ho have
leave a message
Or 
When a people leave a message
Press e.g 3 to mark the message


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Re: [asterisk-users] Attended Transfer question

[Philipp von Klitzing]

Hi!

>> I've been asked to implement the following transfer workflow in an 
>> asterisk system, and I'm not seeing an easy way to do the bolded steps
>> below (steps 4 and 5 for those with a text-only email client):
 
> You could create a dynamic meetme room for the 3 legs and drop out when
> done. Or do it with X static meetme rooms. You could set up a context to
> create the MM room and call the supervisor, connecting them to the room.

Why not just do it with your phone?

* Call the staff member using your 2nd line appearance
* Create a phone-based 3-party conference
* Dissolve the conference
* Transfer the caller to the staff member

If that is not what you want: Look at MeetMe() and ChanSpy().

Philipp


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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[Tzafrir Cohen]

On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote:
>I guess same trouble with Elastix 1.5.2-2.3
>dahdi 2.1.0.4  19
> Asterisk 1.4.25.1
> Digium TDM 2400

That's an analog card.

With an analog trunk, you're not guaranteed to know if the remote CO has
hung up the line.

-- 
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Re: [asterisk-users] voicemail

[Danny Nicholas]

>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
>Subject: [asterisk-users] voicemail

* paraphrasing OP *
Can I add these functions to voicemail?
Some companies have the ability to press 6 to call back the people who have
left a message
Or 
When a people leave a message, Press 3 to mark the message

My answer would be that you can do either or both of these.

I would recommend posting a bounty if it's something you really want and/or
can't figure out.


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[asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release
of Asterisk 1.8.0.

This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

All interested users of Asterisk are encouraged to participate in the 1.8
testing process.  Please report any issues found to the issue tracker,
http://issues.asterisk.org/.  It is also very useful to hear successful test
reports.  Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk.  It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:

 * Secure RTP
 * IPv6 Support
 * Connected Party Identification Support
 * Calendaring Integration
 * A new call logging system, Channel Event Logging (CEL)
 * Distributed Device State using Jabber/XMPP PubSub
 * Call Completion Supplementary Services support
 * Advice of Charge support
 * Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta1

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[bruce bruce]

Well, what about PRI? Why should this stay on? Isn't the native bridge just
a bridge channel that should go down automatically if the actually Dahdi/ZAP
channel is down and there are no SIP channels on either?

Thanks,
Bruce

On Fri, Jul 23, 2010 at 5:09 PM, Tzafrir Cohen wrote:

> On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote:
> >I guess same trouble with Elastix 1.5.2-2.3
> >dahdi 2.1.0.4  19
> > Asterisk 1.4.25.1
> > Digium TDM 2400
>
> That's an analog card.
>
> With an analog trunk, you're not guaranteed to know if the remote CO has
> hung up the line.
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[Maurizio Faccio adinet]

You're right but it do not detect that I hungs on my side of the line.
I think that in some way we are going into a conference in some unwanted 
way with the two dadhi channels and when i hang up both lines stay bridged.
I think that the trouble appears when i dial a number in an analog 
phone, hook quickly (seems like a flash), and dial again.

I am wondering that if I change my lines to a pri I solve this trouble 
but now I do not see clear at all. (my analog telco cannot bring me 
polarity reversal on hang up for signaling

Thank you in advance

Maurizio



El 23/07/2010 06:09 p.m., Tzafrir Cohen escribió:
> On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote:
>
>> I guess same trouble with Elastix 1.5.2-2.3
>> dahdi 2.1.0.4  19
>> Asterisk 1.4.25.1
>> Digium TDM 2400
>>  
> That's an analog card.
>
> With an analog trunk, you're not guaranteed to know if the remote CO has
> hung up the line.
>
>

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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[bruce bruce]

I am having this issue with PRI. But I do not use conference rooms. Our
system is a simple queue and extensions.

-Bruce

On Fri, Jul 23, 2010 at 6:13 PM, Maurizio Faccio adinet <
mauf...@adinet.com.uy> wrote:

> You're right but it do not detect that I hungs on my side of the line.
> I think that in some way we are going into a conference in some unwanted
> way with the two dadhi channels and when i hang up both lines stay bridged.
> I think that the trouble appears when i dial a number in an analog
> phone, hook quickly (seems like a flash), and dial again.
>
> I am wondering that if I change my lines to a pri I solve this trouble
> but now I do not see clear at all. (my analog telco cannot bring me
> polarity reversal on hang up for signaling
>
> Thank you in advance
>
> Maurizio
>
>
>
> El 23/07/2010 06:09 p.m., Tzafrir Cohen escribió:
> > On Fri, Jul 23, 2010 at 05:37:44PM -0300, Maurizio Faccio adinet wrote:
> >
> >> I guess same trouble with Elastix 1.5.2-2.3
> >> dahdi 2.1.0.4  19
> >> Asterisk 1.4.25.1
> >> Digium TDM 2400
> >>
> > That's an analog card.
> >
> > With an analog trunk, you're not guaranteed to know if the remote CO has
> > hung up the line.
> >
> >
>
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Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

[AMARDEEP SINGH]

Do you have working script?

On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese  wrote:

> Maybe you need to read the man page for qpage.  The qpage client can
> send the page to an SNPP server over TCP/IP.
>
> Lyle
>
> AMARDEEP SINGH wrote:
> > Our SMS-gateway is not PSTN accessible.
> >
> > On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese  > > wrote:
> >
> > AMARDEEP SINGH wrote:
> >> Hello All,
> >>
> >> Scenario:
> >> -We use asterisk as voicemail server for our cellular network.
> >> Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway)
> >> through sip.
> >> -Extensions in * are virtual, just for leaving and accessing
> >> voicemail.
> >>
> >> Requirement:
> >> Asterisk to send SMS to cell switch(running SMSC) on reception of
> >> new voicemail.
> >>
> >> Pointers required from Maillist users:
> >> -How can I make * talk to SMSC(ip address:port).
> >> -Anyone using similar topology?
> >> -there are not enough examples/man/maillist of using app_sms(),
> smsq.
> >>
> >> Thanks:
> >> -A
> > qpage?
> >
> >
> >
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> >
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Re: [asterisk-users] Question regarding SMS(), SMSQ, SMSC

[Lyle Giese]

qpage -s snppserver.example.com -p lyle -f lyle test page

AMARDEEP SINGH wrote:
> Do you have working script?
>
> On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese  > wrote:
>
> Maybe you need to read the man page for qpage.  The qpage client can
> send the page to an SNPP server over TCP/IP.
>
> Lyle
>
> AMARDEEP SINGH wrote:
> > Our SMS-gateway is not PSTN accessible.
> >
> > On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese
> mailto:l...@lcrcomputer.net>
> > >> wrote:
> >
> > AMARDEEP SINGH wrote:
> >> Hello All,
> >>
> >> Scenario:
> >> -We use asterisk as voicemail server for our cellular network.
> >> Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway)
> >> through sip.
> >> -Extensions in * are virtual, just for leaving and accessing
> >> voicemail.
> >>
> >> Requirement:
> >> Asterisk to send SMS to cell switch(running SMSC) on
> reception of
> >> new voicemail.
> >>
> >> Pointers required from Maillist users:
> >> -How can I make * talk to SMSC(ip address:port).
> >> -Anyone using similar topology?
> >> -there are not enough examples/man/maillist of using
> app_sms(), smsq.
> >>
> >> Thanks:
> >> -A
> > qpage?
> >
> >
> >
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> >
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> >
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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Ira]

At 02:58 PM 7/23/2010, you wrote:
>The Asterisk Development Team has announced the release of Asterisk 
>1.8.0-beta1.

So being the brave type, I downloaded and installed this onto my 
Asterisk Box. Compiled fine and installed fine, but it didn't work.

I kept getting errors on gosub and none of my DAHDI channels were 
visible. So I went back to 1.6.2.11-beta one and all was well again.

Is there something really basic I missed to get 1.8 to work?

Ira 


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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Philip Prindeville]

  On 7/23/10 6:18 PM, Ira wrote:
> At 02:58 PM 7/23/2010, you wrote:
>> The Asterisk Development Team has announced the release of Asterisk
>> 1.8.0-beta1.
> So being the brave type, I downloaded and installed this onto my
> Asterisk Box. Compiled fine and installed fine, but it didn't work.
>
> I kept getting errors on gosub and none of my DAHDI channels were
> visible. So I went back to 1.6.2.11-beta one and all was well again.
>
> Is there something really basic I missed to get 1.8 to work?
>
> Ira
>
What sort of errors on your Gosub's?


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Re: [asterisk-users] Why does a bridged channel stay open for 4 hours?

[Paul Belanger]

On Fri, Jul 23, 2010 at 6:23 PM, bruce bruce  wrote:
> I am having this issue with PRI. But I do not use conference rooms. Our
> system is a simple queue and extensions.
>
You will then need to enable PRI debugs and check the IE for
disconnect.  The see why Asterisk is not hanging up the channel.

-- 
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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Paul Belanger]

On Fri, Jul 23, 2010 at 8:18 PM, Ira  wrote:
> Is there something really basic I missed to get 1.8 to work?
>
Rather then tell us it did not work, post a debug log showing the issue.

A side from that did you read the UPGRADE.txt and CHANGES file located
in the source directory?

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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Ira]

At 07:08 PM 7/23/2010, you wrote:
>Rather then tell us it did not work, post a debug log showing the issue.
>
>A side from that did you read the UPGRADE.txt and CHANGES file located
>in the source directory?

At least to see if anything seemed to mention gosub or DAHDI. Had 
there been some obvious to me indication of why DAHDI might have 
stopped working I would have kept going, but given there were no 
errors, I'm not that good with linux and it's my business' phone 
system I wasn't willing to leave it down.

Well, looking at messages turned up these, maybe it will help?

WARNING[28505] loader.c: Error loading module 'chan_dahdi.so': 
/usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: 
ast_smdi_interface_unref
WARNING[28505] loader.c: Error loading module 'app_stack.so': 
/usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister
WARNING[28505] loader.c: Error loading module 'func_aes.so': 
/usr/lib/asterisk/modules/func_aes.so: undefined symbol: 
ast_aes_set_decrypt_key
WARNING[28505] loader.c: Error loading module 'app_voicemail.so': 
/usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: 
ast_smdi_mwi_message_destroy

Also, the only way I know to go between 1.62 and 1.8 is "make 
install" which seems like it takes a really long time when the system is down.

Given any hint of what to do that might get dahdi working I'll try 
again. I've been a beta tester a fair amount of my life and 
understand the pitfalls.

I don't figure the gosub issue would be much of a problem, no worse 
than moving extensions.conf from 1.2 to 1.6 but without phone lines, 
it didn't seem like a worthwhile use of my time.

Ira  


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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Richard Kenner]

> WARNING[28505] loader.c: Error loading module 'app_stack.so': 
> /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister

This is the gosub issue.  It's in app_stack.

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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Paul Belanger]

On Fri, Jul 23, 2010 at 10:36 PM, Ira  wrote:
> WARNING[28505] loader.c: Error loading module 'chan_dahdi.so':
> /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol:
> ast_smdi_interface_unref
> WARNING[28505] loader.c: Error loading module 'app_stack.so':
> /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister
> WARNING[28505] loader.c: Error loading module 'func_aes.so':
> /usr/lib/asterisk/modules/func_aes.so: undefined symbol:
> ast_aes_set_decrypt_key
> WARNING[28505] loader.c: Error loading module 'app_voicemail.so':
> /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol:
> ast_smdi_mwi_message_destroy
>
This look to be a build problem with 1.8.  We would need to see a copy
of your config.log and output from 'make install'.  It is possible
your are loading old modules from 1.6 into 1.8.  Check the timestamps
on these modules.
-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] Vocera Comm Badges

[Frank Bulk - iName.com]

I've worked with these before.  They are designed to run a whole hospital
shift, so there should be no worries regarding the battery.

I'm not aware of the server having any kind of SIP support -- I think you
would need to have a PRI trunk to another PBX.  The last time I talked to
them they had their own proprietary codec to deal with the occasional packet
loss of Wi-Fi, and the codec was encrypted, too.

They really work -- just hit the button and say "Call Bob Smith" and they
call Bob.  

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Friday, July 23, 2010 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Vocera Comm Badges

Hi!

> I´ve seen them at trade shows, I think I remember it being proprietary.
> What about using Dect handsets?

That Star Trek device has always interested me. Too bad they chose WiFi 
over DECT, though.

Vocera badge:
* WLAN b/g
* Talktime 2-2.5 hours, standby 20-27 hours
* headset jack
* OLED display (why don't they ever show this on the pics)
* Linux based

Back side:
http://farm4.static.flickr.com/3412/3496101115_32319840a5.jpg

System:
* Windows server
* Nuance ASR & biometrics & Dictaphone
* Dialogic T1/ISDN/Analog cards
* SIP interface
* iPhone app

I am also curious to hear some user reports, in particular about battery 
performance: Does it last one entire working day, or does it need to be 
exchanged (once?) per day? What is the smallest feasible installation? 
What speech protocol does the badge use to talk to the server?

"First quarter of 2010 was the third consecutive profitable quarter for 
the company. In addition, since the beginning of 2010, the company has 
announced the addition of 22 new employees across North America and the 
United Kingdom."

Philipp


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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Jose P. Espinal]

Just my 2 cents,

Try this to see if it helps:

- Try removing the Dahdi modules loaded into your kernel
- Run /sbin/depmod
- Reinsert the modules using modprobe [module name]
- Restart Asterisk



Ira wrote:
> At 07:08 PM 7/23/2010, you wrote:
>   
>> Rather then tell us it did not work, post a debug log showing the issue.
>>
>> A side from that did you read the UPGRADE.txt and CHANGES file located
>> in the source directory?
>> 
>
> At least to see if anything seemed to mention gosub or DAHDI. Had 
> there been some obvious to me indication of why DAHDI might have 
> stopped working I would have kept going, but given there were no 
> errors, I'm not that good with linux and it's my business' phone 
> system I wasn't willing to leave it down.
>
> Well, looking at messages turned up these, maybe it will help?
>
> WARNING[28505] loader.c: Error loading module 'chan_dahdi.so': 
> /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: 
> ast_smdi_interface_unref
> WARNING[28505] loader.c: Error loading module 'app_stack.so': 
> /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister
> WARNING[28505] loader.c: Error loading module 'func_aes.so': 
> /usr/lib/asterisk/modules/func_aes.so: undefined symbol: 
> ast_aes_set_decrypt_key
> WARNING[28505] loader.c: Error loading module 'app_voicemail.so': 
> /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: 
> ast_smdi_mwi_message_destroy
>
> Also, the only way I know to go between 1.62 and 1.8 is "make 
> install" which seems like it takes a really long time when the system is down.
>
> Given any hint of what to do that might get dahdi working I'll try 
> again. I've been a beta tester a fair amount of my life and 
> understand the pitfalls.
>
> I don't figure the gosub issue would be much of a problem, no worse 
> than moving extensions.conf from 1.2 to 1.6 but without phone lines, 
> it didn't seem like a worthwhile use of my time.
>
> Ira  
>
>
>   

-- 
Jose P. Espinal
http://www.eslackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs


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Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!

[Ira]

At 08:13 PM 7/23/2010, you wrote:
>This look to be a build problem with 1.8.  We would need to see a copy
>of your config.log and output from 'make install'.  It is possible
>your are loading old modules from 1.6 into 1.8.  Check the timestamps
>on these modules.

So I went back to the 1.8 folder, did make clean and then make, then 
I deleted the contents of  usr/lib/asterisk/modules and then ran "make install"

Still the same errors. My config.log is here, 
www.behmorthing.com/Downloads/beta/config.zip .  I can see errors in 
it, but I'm sorry to say I have no clue what it all means.

Is there an easy way to move between 2 versions of Asterisk on one machine?

Ira  


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[asterisk-users] getting some segmentation faults with 1.8

[covici]

I downloaded the latest 1.8 (27922) but got some segmentation faults.
The first one was when it loaded cdr_odb, and so I changed menuselect
not to compile that one, but the second one was when it tried to load
chan_agent and so I stopped there to see if anyone else was seeing
this.  The agents.conf is all commented out except for [general] .

Anyone know what is happening?

Thanks.

P.S. I deleted the modules directory before installing.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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