[asterisk-users] Mexican ITSP needed

[Michiel van Baak]

Hey all,

I was wondering if anyone knows about a Mexican ITSP I can connect to to
route calls from and to my * boxen.

If it matters: I'm located in The Netherlands and one of our customers
is in Mexico so if we need a Mexican presence that is not an issue.

Thanks.

-- 

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

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


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Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

[Andrew Thomas]

Why are you putting semi-colons at the end of every line?  The dialplan
isn't written in PHP ;).



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L.
Kline
Sent: 15 July 2009 23:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mark Michelson wrote:
> 
> You need to set a call-limit for the SIP peer. Device state
calculation for a 
> SIP peer is predicated on both the call-limit and busylevel. Let's say
that you 
> were to have a call-limit of 2, but no busylevel set. These are the
device 
> states reported for the peer based on the number of calls currently
handled:


Hi Mark.  Thanks for your explanation of these parameters.

I should have posted my configurations.  I double-checked the contents
of sip.conf and I have this.  The 'subscribecontext' was added for
testing, per the other reply I got for my question.

;
; Settings common to all devices on our system
;
[basic-options](!)
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes

;
; Standard desksets here
;
[lan-deskset](!,basic-options)
context=sip-deskset
notifyringing = yes
notifyhold = yes
limitonpeers = yes
call-limit=99

[6668](lan-deskset)
secret=mysecret
callerid="Matts SIP" <6668>
username=Barry's IP450
call-limit=32
busylevel=1
subscribecontext=hint-context


My hint-context is:

[hint-context]

exten => 6668,hint,SIP/6668;


I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
Here is the CLI output:

[Jul 15 18:40:15] -- Executing [6...@sip-deskset:1]
NoOp("SIP/-0955ecc8", "SIP/6668 has state NOT_INUSE") in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:2]
NoOp("SIP/-0955ecc8", "SIP/ has state NOT_INUSE") in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:3]
ExecIf("SIP/-0955ecc8", "0?Busy(10)") in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:4]
Dial("SIP/-0955ecc8", "SIP/6668") in new stack


And here is sip show inuse:

corp-asterisk*CLI> sip show inuse
* User name   In use  Limit
6668  1   32
6667  0   99
  1   99
* Peer name   In use  Limit
6668  1/1/0   32
6667  0/0/0   99
  0/0/0   99


For completeness, here is the dialplan that's producing this:

exten => 6668,1,NoOp(SIP/${EXTEN} has state
${DEVICE_STATE(SIP/${EXTEN})});
exten => 6668,n,NoOp(SIP/ has state ${DEVICE_STATE(SIP/)});
exten =>
6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}="BUSY"]?Busy(10));
exten => 6668,n,Dial(SIP/${EXTEN});


-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFKXlwaCFu3bIiwtTARAkRpAJ4+2WF9qrIwrC3Kdpwd0YAOm/5S1wCfUR1T
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[asterisk-users] Sending things to Jabber but not within an extension

[Phil Reynolds]

I have set my Asterisk server up to connect to my Jabber server and  
send messages with the caller ID details in them to the recipients of  
incoming calls - this is working very nicely.

There are a few other things I can think of right now that I would  
like to send to Jabber but as yet I do not know whether they are  
possible. They are:

(a) a count of messages in a voicemail box - triggered when the user  
connects to Jabber.

(b) a notification that a voicemail was left - triggered when the  
caller leaving the message hangs up.

(c) a notification that a specific SIP peer has become unreachable -  
or better still, has been unreachable for five minutes.

As I say, I do not know if any of these things are possible, nor how  
to do them if they are. I have looked at what is on voip-info.org and  
what the book has but have not seen anything that seems relevant.

Of course, I may be right that these are not possible at present, or I  
may have missed or misunderstood how to do it. I am on version  
1.4.21.2~dfsg-1-pmr-2 - which is from Debian stable plus a locally  
added patch.

Thanks in advance to anyone who can advise, even if just to confirm  
that it is not possible yet.

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95



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Re: [asterisk-users] Generic question about PBX PRI installs

[Dale Noll]

Jerry Geis wrote:



The PBX guy "seems" to always complain about how he has MANY options
and thats not enough information...

What else am I supposed to supply this person. Are they not the PBX 
expert?...


Anyway as example. the last customer I told the above information. He 
set up the PBX
and I can make 4 digit calls successfully, 7 digit and long distance are 
not successful.
They are hitting some error condition that the call is going to the 
switch board.

So the connection is working just not completely working.
This is connecting to a nortel 1000 pbx.

I dont know anything about a nortel switch to them what to change.

What should I be supplying to these PBX guys to get the installs going 
smoother and quicker?


I have connected my Nortel Meridian to * via PRI successfully(?). 
Nortel has what is called BARS/NARS.  The ability to dial numbers to 
access certain routes is based on the Network Class of Service (NCOS) of 
a telephone or trunk.  I would suspect that the user has the trunk set 
with a NCOS to low to access the outbound routes.  Also, tandem calls 
from the Tie line to the outside trunks will need to dial the proper 
access code to activate BARS/NARS. (Ie.  If a Nortel user needs to dial 
9, so does the call from the tie trunk.)


There is also the option, instead of using the BARS/NARS access code, 
the tie can use the route access code (ACOD) directly.


As far as the PRI setup, I have found that I prefer QSIG (if the Nortel 
system release supports it) because if will properly pass all the CLID 
info between systems, NI2 did not.


I hope this helps.

Dale



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[asterisk-users] AGI to announce temperature from weather.com XML file

[Trevor Hammonds]

I would like to have the ability to have Asterisk announce the temperature
-- not using TTS -- within the dialplan.  

For a non-Asterisk project, I have a cron job that periodically pulls down
an XML file from weather.com containing local weather data (TWC's user
agreement requires that data be cached locally).  Using sed, I also create a
text file that contains only the numeric value of the current temperature,
created from that XML file (e.g. 65 in the XML file becomes a
text file with 65 as its only contents).  

I am hoping someone on the list has an example of a lightweight AGI script
that I may modify to either read the simple text file and set a dialplan
variable to the current temperature, or hopefully a more-sophisticated one
which will parse the XML file to set the dialplan variable.  

The end goal is to have Asterisk play the speech files "temperature" "sixty"
"five" "degrees" or the equivalent non-English files per the channel's
current language setting.  

Thank you.  Any assistance will be greatly appreciated.  

Sincerely,
Trevor Hammonds



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[asterisk-users] Struggling with Macros and "s" Extension

[Alan Lord (News)]

Hi all,

I'm sure this has been done before but I just can't figure it out.

On my * box I have a simple IVR:

[tolc_menu] ; Welcome and information to callers
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Background(welcome-to-tolc) ; Say Hello
exten => s,n,Wait(1)
exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter 
extension number if known, or
exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
exten => s,n,WaitExten(5)
exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)

exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})

exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})

.
.
.

Hopefully you'll see that the caller can either enter an extension 
number or wait. If they wait, we use macro-belllord:

[macro-belllord]
exten => s,1,Dial(${ARG1},20,t)
exten => s,n,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the 
voicemail context, ${ARG2} is the mailbox number to dial
exten => s-NOANSWER,n,Hangup()

exten => s-BUSY,1,Voicemail(${ar...@business,b)
exten => s-BUSY,n,Hangup()

exten => _s-.,1,Goto(s-NOANSWER,1)

The Vars ALANL and ALANB are:

ALANL=SIP/101
ALANB=IAX2/alanb/202

If I call in, dial the extension (say 101) and connect, then the Link 
Event on the AMI port (and in CDRs) correctly displays *both* numbers of 
the connection.

For that scenario we use macro-call_extention:

[macro-call_extension]
exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.

exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)

exten => s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)

exten => _s-.,1,Goto(s-NOANSWER,1)


If however I wait and let macro-belllord do it's stuff. I only ever see 
"s" as the called party's number.

I really need to know what that extension number is.

Could someone help me and show how I can rejig this? It was suggested to 
do something with ${MACRO_EXTEN} but I can't get it at all...

Many thanks in advance.

Alan


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Re: [asterisk-users] advices on how to debridge/rebridge a call?

[Leif Madsen]

Sebastian Maz wrote:
> this is what I'm trying to accomplish:
> 
>  - receiving an inbound call from A
>  - dialing another number (B)
>  - bridge A and B
>  - every x minutes, debridge A and B, and bridge A with C (SIP call to 
> an platform that is gonna play an ad)
>  - rebridge A and B
> 
> Any advice on how to do this? Dial cmd with the use of the M or G option?
> Asterisk Manager? AGI?
> 
> Thanks for your ideas

There is the Bridge() application in Asterisk 1.6.x that may be useful. You 
could probably use some sort of script that connects to the manager, and uses 
that application to move the bridges around.

Leif Madsen
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] how to enable dial to a...@asterisk.blurb.com

[Leif Madsen]

John A. Sullivan III wrote:
> On Wed, 2009-07-15 at 14:34 +1000, Alex Samad wrote:
>> The subject line says it all how do I enable this style of call.
>> Pointers to the dns setup and asterisk setup would be great
> 
> If I understand what you are seeking, you can try these URIs:
> 
> http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
> http://www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/


Great posts (especially the blyon.com one). If you're looking for a DNS 
provider 
(assuming you run your own domain name, but don't have your own DNS servers), 
then I've found www.editdns.net is great for this kind of thing. They allow you 
to setup SRV records (unlike some of the other free DNS hosters I've used).

Note that I'm not affiliated with editdns.net in any way -- just found them 
when 
trying to do the same thing :)

Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] AGI to announce temperature from weather.com XMLfile

[Andrew Thomas]

I have just the thing in PHP.

Drop me a personal e-mail and I'll whiz it over.

Andrew Thomas
Technical Services Manager
a...@datavox.co.uk
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF 



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Trevor
Hammonds
Sent: 16 July 2009 12:50
To: 'Asterisk Users List'
Subject: [asterisk-users] AGI to announce temperature from weather.com
XMLfile

I would like to have the ability to have Asterisk announce the
temperature
-- not using TTS -- within the dialplan.  

For a non-Asterisk project, I have a cron job that periodically pulls
down
an XML file from weather.com containing local weather data (TWC's user
agreement requires that data be cached locally).  Using sed, I also
create a
text file that contains only the numeric value of the current
temperature,
created from that XML file (e.g. 65 in the XML file becomes a
text file with 65 as its only contents).  

I am hoping someone on the list has an example of a lightweight AGI
script
that I may modify to either read the simple text file and set a dialplan
variable to the current temperature, or hopefully a more-sophisticated
one
which will parse the XML file to set the dialplan variable.  

The end goal is to have Asterisk play the speech files "temperature"
"sixty"
"five" "degrees" or the equivalent non-English files per the channel's
current language setting.  

Thank you.  Any assistance will be greatly appreciated.  

Sincerely,
Trevor Hammonds



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Re: [asterisk-users] AGI to announce temperature from weather.com XML file

[John Novack]

Check this one out.
developed for AstLinux, it ought to be close to what you want.
depending on your version, you may need to modify sound file references

http://lonnie.abelbeck.com/astlinux/info/weather.php

John Novack


Trevor Hammonds wrote:
> I would like to have the ability to have Asterisk announce the temperature
> -- not using TTS -- within the dialplan.  
>
> For a non-Asterisk project, I have a cron job that periodically pulls down
> an XML file from weather.com containing local weather data (TWC's user
> agreement requires that data be cached locally).  Using sed, I also create a
> text file that contains only the numeric value of the current temperature,
> created from that XML file (e.g. 65 in the XML file becomes a
> text file with 65 as its only contents).  
>
> I am hoping someone on the list has an example of a lightweight AGI script
> that I may modify to either read the simple text file and set a dialplan
> variable to the current temperature, or hopefully a more-sophisticated one
> which will parse the XML file to set the dialplan variable.  
>
> The end goal is to have Asterisk play the speech files "temperature" "sixty"
> "five" "degrees" or the equivalent non-English files per the channel's
> current language setting.  
>
> Thank you.  Any assistance will be greatly appreciated.  
>
> Sincerely,
> Trevor Hammonds
>
>
>
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>   

-- 
Dog is my co-pilot


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Re: [asterisk-users] Sending things to Jabber but not within anextension

[Danny Nicholas]

Each of these should be do-able either through dialplan snippets, cron jobs
or AGI's.  (a) would be a dialplan snippet (b) would be a dialplan snippet
(c) would require an AGI or cron to monitor how long the peer has been out
of service.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Reynolds
Sent: Thursday, July 16, 2009 5:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sending things to Jabber but not within
anextension

I have set my Asterisk server up to connect to my Jabber server and  
send messages with the caller ID details in them to the recipients of  
incoming calls - this is working very nicely.

There are a few other things I can think of right now that I would  
like to send to Jabber but as yet I do not know whether they are  
possible. They are:

(a) a count of messages in a voicemail box - triggered when the user  
connects to Jabber.

(b) a notification that a voicemail was left - triggered when the  
caller leaving the message hangs up.

(c) a notification that a specific SIP peer has become unreachable -  
or better still, has been unreachable for five minutes.

As I say, I do not know if any of these things are possible, nor how  
to do them if they are. I have looked at what is on voip-info.org and  
what the book has but have not seen anything that seems relevant.

Of course, I may be right that these are not possible at present, or I  
may have missed or misunderstood how to do it. I am on version  
1.4.21.2~dfsg-1-pmr-2 - which is from Debian stable plus a locally  
added patch.

Thanks in advance to anyone who can advise, even if just to confirm  
that it is not possible yet.

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95



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Re: [asterisk-users] AGI to announce temperature from weather.com XML file

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Trevor Hammonds wrote:

> I am hoping someone on the list has an example of a lightweight AGI script
> that I may modify to either read the simple text file and set a dialplan
> variable to the current temperature, or hopefully a more-sophisticated one
> which will parse the XML file to set the dialplan variable.  

I think that the 'FILE' function will do what you're looking for.

Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

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=YLth
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[asterisk-users] Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1

[hutx]

I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax 
(.tiff) from the first
asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an 
INVITE with audio G.711. Asterisk2
accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to 
Asterisk1. But, Asterisk1 responds with
"488 not acceptable here". I double check t38pt_udptl = yes in my sip.conf. Why 
not Asterisk1 can not accept the Re-INVITE
with T.38 SDP? What do I miss?



dev10*CLI> fax show version
Fax For Asterisk Components:
dev10*CLApplications: 1.6.1_1.0.11
dev10*CLDigium Fax T.38 Driver: 1.6.1_1.0.9 (optimized for i686_32)
dev10*CLDigium Fax G.711 Driver: 1.6.1_1.0.9 (optimized for i686_32)

--
.call file

Channel: SIP/1...@outbound-calls
MaxRetries: 3
WaitTime: 30
Set: LOCALSTATIONID=2
Set: LOCALHEADERINFO=T38 fax
Set: T38CALL=1
Set: T38TXDETECT=yes
CallerID: 123456
Context: fax-tx
Extension: send
priority:1


---
sip.conf

[general]
context=default 
allowoverlap=no 
udpbindaddr=0.0.0.0 
tcpenable=no
tcpbindaddr=0.0.0.0 
srvlookup=yes   
disallow=all
allow=ulaw  
t38pt_udptl = yes   



extensions.conf

[fax-rx]
exten => receive,1,NoOp( FAX RECEIVE )
exten => receive,n,Set(GLOBAL(FAXCOUNT)=${GLOBAL(FAXCOUNT)}+1)
exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten => receive,n,Set(FAXFILE=fax-${FAXCOUNT}-rx.tif)
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten => receive,n,NoOp( SETTING FAXOPT )
exten => receive,n,Set(FAXOPT(ecm)=yes)
exten => receive,n,Set(FAXOPT(headerinfo)=MY FAXBACK RX)
exten => receive,n,Set(FAXOPT(localstationid)=1234567890)
exten => receive,n,Set(FAXOPT(maxrate)=14400)
exten => receive,n,Set(FAXOPT(minrate)=2400)
exten => receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten => receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten => receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten => receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten => receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten => receive,n,NoOp( RECEIVING FAX : ${FAXFILE} )
exten => receive,n,ReceiveFAX(/home/sip/fax/${FAXFILE})

[fax-tx]
exten => send,1,NoOp( SENDING FAX )
exten => send,n,Wait(6)
exten => send,n,Set(GLOBAL(FAXCOUNT)=1)
;exten => send,n,Set(GLOBAL(FAXCOUNT)= ${GLOBAL(FAXCOUNT)}+1)
exten => send.,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten => send,n,Set(FAXFILE=test.tif)
; Set FAXOPTs
exten => send,n,NoOp( SETTING FAXOPT )
exten => send,n,Set(FAXOPT(ecm)=yes)
exten => send,n,Set(FAXOPT(headerinfo)=Fax from ${GLOBAL(LASTFAXCALLERNAME)} at 
${GLOBAL(LASTFAXCALLERNUM)} was received.)
exten => send,n,Set(FAXOPT(localstationid)=1234567890)
exten => send,n,Set(FAXOPT(maxrate)=14400)
exten => send,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten => send,n,NoOp( SENDING FAX : ${FAXFILE} )
exten => send,n,SendFAX(/home/sip/fax/${FAXFILE},d)


[default]
exten => _X.,1,NoOp( FAX DETECTED )
exten => _X.,n,Goto(fax-rx,receive,1)

--

The SIP trace is

#
U 2009/07/15 22:30:11.588436 74.13.233.143:5060 -> 209.167.0.151:5060
  INVITE sip:1...@209.167.0.151 SIP/2.0..Via: SIP/2.0/UDP 
74.13.233.143:5060;branch=z9hG4bK092e48ce;rport..Max-Forwards: 70..From
  : "123456" ;tag=as74992a24..To: 
..Contact: ..Call-I
  D: 422fd4375fe79a5977e891870f5cc...@74.13.233.143..cseq: 102 
INVITE..User-Agent: Asterisk PBX 1.6.1.1..Date: Wed, 15 Jul 2009 2
  2:30:11 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY..Supported: replaces, timer..Content-Type: appl
  ication/sdp..Content-Length: 265v=0..o=root 1425900082 1425900082 IN IP4 
74.13.233.143..s=Asterisk PBX 1.6.1.1..c=IN IP4 74
  .13.233.143..t=0 0..m=audio 18452 RTP/AVP 0 101..a=rtpmap:0 
PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=si
  lenceSupp:off - - - -..a=ptime:20..a=sendrecv..   
 
#
U 2009/07/15 22:30:11.723006 209.167.0.151:5060 -> 74.13.233.143:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 
74.13.233.143:5060;branch=z9hG4bK092e48ce;received=74.13.233.143;rport=5060..From:
 "123456
  " ;tag=as74992a24..To: 
..Call-ID: 422fd4375fe79a5977e891870f5cc...@74.13.233.
  143..CSeq: 102 INVITE..Server: Asterisk PBX 1.6.1.1..Allow: INVITE, ACK, 
CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Suppor
  ted: replaces, timer..Contact: ..Content-Length: 
0 
#
U 2009/07/15 22:30:11.730205 209.167.0.151:5060 -> 74.13.233.143:5060
  SIP/2.0 

Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.

[sean darcy]

On Wed, Jul 15, 2009 at 5:23 PM, Wayne wrote:
> Hi all,
> Just a quickie to say that this has been solved now - real simple -
> downloaded '*current*' rather than the versions from the home page of
> Astrisk.org. (didn't realise there was a 'current' version tbh.
>
> Anyways - I don't get Asterisk seg faulting now when hammering the
> speaker button on my cisco phones :)
>
> Interestingly - I've got another query - but will post another question
> when I've had chance to play more.
>
> Cheers
> Wayne.
>
> Wayne wrote:
>> Hi all,
>> I've just built a new installation of CentOS release 5.3 (Final) and
>> have installed both
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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>

Huh?

http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz

is not the same as

http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz ?

Their sha1 files are identical.

sean

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Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

[Mark Michelson]

Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Mark Michelson wrote:
>> You need to set a call-limit for the SIP peer. Device state calculation for 
>> a 
>> SIP peer is predicated on both the call-limit and busylevel. Let's say that 
>> you 
>> were to have a call-limit of 2, but no busylevel set. These are the device 
>> states reported for the peer based on the number of calls currently handled:
> 
> 
> Hi Mark.  Thanks for your explanation of these parameters.
> 
> I should have posted my configurations.  I double-checked the contents
> of sip.conf and I have this.  The 'subscribecontext' was added for
> testing, per the other reply I got for my question.
> 
> ;
> ; Settings common to all devices on our system
> ;
> [basic-options](!)
> type=friend
> host=dynamic
> canreinvite=no
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> qualify=yes
> 
> ;
> ; Standard desksets here
> ;
> [lan-deskset](!,basic-options)
> context=sip-deskset
> notifyringing = yes
> notifyhold = yes
> limitonpeers = yes
> call-limit=99
> 
> [6668](lan-deskset)
> secret=mysecret
> callerid="Matts SIP" <6668>
> username=Barry's IP450
>   call-limit=32
> busylevel=1
> subscribecontext=hint-context
> 
> 
> My hint-context is:
> 
> [hint-context]
> 
> exten => 6668,hint,SIP/6668;
> 
> 
> I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
> Here is the CLI output:
> 
> [Jul 15 18:40:15] -- Executing [6...@sip-deskset:1]
> NoOp("SIP/-0955ecc8", "SIP/6668 has state NOT_INUSE") in new stack
> [Jul 15 18:40:15] -- Executing [6...@sip-deskset:2]
> NoOp("SIP/-0955ecc8", "SIP/ has state NOT_INUSE") in new stack
> [Jul 15 18:40:15] -- Executing [6...@sip-deskset:3]
> ExecIf("SIP/-0955ecc8", "0?Busy(10)") in new stack
> [Jul 15 18:40:15] -- Executing [6...@sip-deskset:4]
> Dial("SIP/-0955ecc8", "SIP/6668") in new stack
> 
> 
> And here is sip show inuse:
> 
> corp-asterisk*CLI> sip show inuse
> * User name   In use  Limit
> 6668  1   32
> 6667  0   99
>   1   99
> * Peer name   In use  Limit
> 6668  1/1/0   32
> 6667  0/0/0   99
>   0/0/0   99
> 
> 
> For completeness, here is the dialplan that's producing this:
> 
> exten => 6668,1,NoOp(SIP/${EXTEN} has state ${DEVICE_STATE(SIP/${EXTEN})});
> exten => 6668,n,NoOp(SIP/ has state ${DEVICE_STATE(SIP/)});
> exten => 6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}="BUSY"]?Busy(10));
> exten => 6668,n,Dial(SIP/${EXTEN});
> 
> 

Thanks for the config info. I have a couple of suggestions for fixes.

1. Try changing the type in [basic-options] from friend to peer. I've found 
that 
device state reporting for outbound calls (from the perspective of the phone) 
tends to be more accurate with this type.

2. If for some odd reason number 1 either doesn't sound appealing to you or 
doesn't work, then try moving the limitonpeers=yes option from your 
[basic-options] section to the [general] section.

No, neither of these ideas actually make any real sense to me, but they are 
based on behavior that I have witnessed with my Asterisk setup in my office.

Mark Michelson

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Re: [asterisk-users] Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1

[Kevin P. Fleming]

hutx wrote:
> I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax 
> (.tiff) from the first
> asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates 
> an INVITE with audio G.711. Asterisk2
> accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to 
> Asterisk1. But, Asterisk1 responds with
> "488 not acceptable here". I double check t38pt_udptl = yes in my sip.conf. 
> Why not Asterisk1 can not accept the Re-INVITE
> with T.38 SDP? What do I miss?

You need to post a console log from the Asterisk console of this problem
occurring, with 'core set verbose 10', 'core set debug 10' and 'sip set
debug on' (and ensure that the 'debug' logger level is activated for the
console log channel in logger.conf).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] H323 situation

[Luis Silva]

Hi all,

I have this installation:

Asterisk 1.6.1.1  with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. 

I have a  problem that is, when a call comes from H323 and goes to a Sip
phone the asterisk sends two rtp streams to the sip. I checked this with
tcpdump, save the payload (voice is in G711u), one is the ringing indication
and the other is the voice coming from the user in h323 side. And worst they
go to the same port. This causes that in the sip phone there are problems,
when the call is answered sometimes we get the riging indication, others a
mix of the two with very bad sound quality and others(few) a god audio call.


The outgoing calls from sip to H323 are ok.

I also tested an incoming call from a dahdi channel and from here everything
is ok, only one rtp stream and a good call.

 

By the way I had other problem that I fixed, but don't know if it was in the
best way. 

The h323 box is a Cisco AS5300 (or 5350?) and when I was making outgoing
calls the AS disconnected all of them after 10 sec. 

 I investigated I noticed that the AS as a limitation to the G711 payload to
20 ms, and asterisk was using 150 ms. I resolve this changing in frame.c the
codec value and recompile asterisk. There is simpler way to do this? Like
changing values in codec.conf?...

 

Regards

LS 

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[asterisk-users] Voicemail login incorrect

[Zaheer Master]

Hi all,
I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled
voicemail in the extensions area, and set the default password. However,
every time I try to log in with a mailbox and password, I get the message
"login incorrect". I've tried changing the voicemail password, and also
disabling and re-enabling the voicemail feature. What else can I do to set
up the voicemail? Also, I've left the VM Context as "default" and the
mailbox is "101".

Thanks!

--Zaheer


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Re: [asterisk-users] Suggestions for web based soft phones

[Brian]

Hi Zeeshan,

You might want to take a look at our solution here: http://www.flashsip.com/
We do the customization of the software for our clients on demande.

Best regards,

Brian

On Sat, Jul 11, 2009 at 2:55 AM, Zeeshan Zakaria  wrote:

> For a while now I've been looking for a good web based soft phone solution,
> but so far no luck. A few solutions which I've tried, both Java based and
> Flash based, either don't work, or had bad sound quality. I need something
> which I could put on my productions server for my clients.
>
> Seems like good web based solutions are all paid ones, nobody is giving it
> for free. Any ideas, suggestions whom to go with?
>
> Thanks
>
> --
> Zeeshan A Zakaria
>
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Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mark Michelson wrote:

> Thanks for the config info. I have a couple of suggestions for fixes.
> 
> 1. Try changing the type in [basic-options] from friend to peer. I've found 
> that 
> device state reporting for outbound calls (from the perspective of the phone) 
> tends to be more accurate with this type.
> 
> 2. If for some odd reason number 1 either doesn't sound appealing to you or 
> doesn't work, then try moving the limitonpeers=yes option from your 
> [basic-options] section to the [general] section.
> 
> No, neither of these ideas actually make any real sense to me, but they are 
> based on behavior that I have witnessed with my Asterisk setup in my office.
> 
> Mark Michelson

I'll give these a try and see if they help.  At this point, I'd be
willing to slaughter a goat and place its entrails on the keyboard if I
thought it would help.

Barry

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

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Jw5YOgQllm1+wbq2wThh4Wg=
=eE0m
-END PGP SIGNATURE-

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[asterisk-users] T38 negotiation, the last step !

[Xavier Cardil]

Hi, I've managed to get HYLAFAX>T38MODEM->ASTERISK>CISCOAS5400
working, but when they are negotiating asterisk drops a message telling
"Unknown RTP codec 96 received from gateway" Do somebody know how to fix it
?

Thank you !



<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ]
[SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
[Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec
96 received from '192.168.3.163'
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
[Jul 16 17:50:40] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec
96 received from '192.168.3.163'
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
[Jul 16 17:50:41] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec
96 received from '192.168.3.163'
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
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Re: [asterisk-users] Voicemail login incorrect

[John A. Sullivan III]

On Thu, 2009-07-16 at 10:57 -0400, Zaheer Master wrote:
> Hi all,
> I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled
> voicemail in the extensions area, and set the default password. However,
> every time I try to log in with a mailbox and password, I get the message
> "login incorrect". I've tried changing the voicemail password, and also
> disabling and re-enabling the voicemail feature. What else can I do to set
> up the voicemail? Also, I've left the VM Context as "default" and the
> mailbox is "101".

If you set your Asterisk console to a verbose mode, what password do you
see passed to the voicemail application? We recently noticed 3CX
softphone users with multiple options set for DTMF were sending
duplicate DTMF signals to our voicemail resulting in the same problems
you are seeing, e.g., 1234 would be sent as 11223344.  Just a guess -
John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Voicemail login incorrect - SOLVED

[Zaheer Master]

Thanks for the reply John. In the voicemail.conf file there were two extra
[] creating a NULL context. Removing those extra brackets fixed the problem.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Thursday, July 16, 2009 12:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: 'AsteriskNOW Discussion'
Subject: Re: [asterisk-users] Voicemail login incorrect

On Thu, 2009-07-16 at 10:57 -0400, Zaheer Master wrote:
> Hi all,
> I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have
enabled
> voicemail in the extensions area, and set the default password. However,
> every time I try to log in with a mailbox and password, I get the message
> "login incorrect". I've tried changing the voicemail password, and also
> disabling and re-enabling the voicemail feature. What else can I do to set
> up the voicemail? Also, I've left the VM Context as "default" and the
> mailbox is "101".

If you set your Asterisk console to a verbose mode, what password do you
see passed to the voicemail application? We recently noticed 3CX
softphone users with multiple options set for DTMF were sending
duplicate DTMF signals to our voicemail resulting in the same problems
you are seeing, e.g., 1234 would be sent as 11223344.  Just a guess -
John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] T38 negotiation, the last step !

[Kevin P. Fleming]

Xavier Cardil wrote:
> Hi, I've managed to get HYLAFAX>T38MODEM->
> ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk
> drops a message telling "Unknown RTP codec 96 received from gateway" Do
> somebody know how to fix it ? 

There's nothing to fix; the gateway sent an expected RTP packet, which
Asterisk dropped. If your FAX works, then you can ignore this.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.

[Jonathan Thurman]

> Huh?
>
>
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz
>
> is not the same as
>
>
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz?
>
> Their sha1 files are identical.
>
> sean
>
>
I believe he means that:


http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz

is not the same as

svn checkout http://svn.digium.com/svn/asterisk/branches/1.6.0



Which is true as there are lots of things that have been fixed in the
Subversion repo.

-Jonathan
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Re: [asterisk-users] Mexican ITSP needed

[Carlos Chavez]

Try http://www.inext.com.mx they can provide DIDs in several cities in
Mexico.

On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote:
> Hey all,
> 
> I was wondering if anyone knows about a Mexican ITSP I can connect to to
> route calls from and to my * boxen.
> 
> If it matters: I'm located in The Netherlands and one of our customers
> is in Mexico so if we need a Mexican presence that is not an issue.
> 
> Thanks.
> 
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] T38 negotiation, the last step !

[Miguel Molina]

Xavier Cardil escribió:
> Hi, I've managed to get HYLAFAX>T38MODEM->
> ASTERISK>CISCOAS5400 working, but when they are negotiating 
> asterisk drops a message telling "Unknown RTP codec 96 received from 
> gateway" Do somebody know how to fix it ? 
>
> Thank you !
>
>
>
> << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] 
> [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP 
> codec 96 received from '192.168.3.163'
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> [Jul 16 17:50:40] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP 
> codec 96 received from '192.168.3.163'
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> [Jul 16 17:50:41] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP 
> codec 96 received from '192.168.3.163'
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
It looks like the Cisco gateway is trying to negotiate or send some 
video codec or something which asterisk does not recognize. According to 
the spec (http://www.iana.org/assignments/rtp-parameters), The codecs 96 
to 127 are dynamic RTP payloads that are negotiated with another 
protocols to define them. If everything else on your fax session works 
fine, this message shouldn't be a problem. I think this is much like the 
X-lite 126 codec type message:

[Jul 16 11:35:57] NOTICE[10989]: rtp.c:1287 ast_rtp_read: Unknown RTP 
codec 126 received from '0.0.0.0'

And doesn't pose a problem anyway.

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57)3138873587 


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Re: [asterisk-users] T38 negotiation, the last step !

[Xavier Cardil]

Hi Kelvin, thank you for your response, well in fact it is not working but
that's only a NOTICE, not an error. Warnings comes after that and the fax is
not sent. Take a look at the last lines of this output :


<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-01ba5638]
IVR3*CLI> debug channel
allSIP/GWCISCO5400O-01ba5638
SIP/T38modem-01ba2178
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-01ba5638]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638]
<< [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ]
[SIP/GWCISCO5400O-01ba5638]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638]
[Jul 16 18:20:20] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec
96 received from '192.168.3.163'
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638]
[Jul 16 18:20:21] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec
96 received from '192.168.3.163'
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638]
[Jul 16 18:20:21] NOTICE[2846]: rtp.c:1739 ast_rtp_read: Unknown RTP codec
96 received from '192.168.3.163'
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-01ba5638]
[Jul 16 18:20:30] WARNING[2815]: chan_sip.c:3075 retrans_pkt: Maximum
retries exceeded on transmission
1af1c92e-9270-de11-8264-001517bb9...@ivr3for seqno 1 (Critical
Response) -- See doc/sip-retransmit.txt.
[Jul 16 18:20:30] WARNING[2815]: chan_sip.c:3102 retrans_pkt: Hanging up
call 1af1c92e-9270-de11-8264-001517bb9...@ivr3 - no reply to our critical
packet (see doc/sip-retransmit.txt).

Do you have an idea of what is happening ? I sniffed UDP traffic on port
5060 and I get :
INVITE SDP ( t38 ) -->
TRYING <--
NOT ACCEPTABLE HERE<
ACK>
BYE<---
200 OK>

Thanks  for your help.



On Thu, Jul 16, 2009 at 6:17 PM, Kevin P. Fleming wrote:

> Xavier Cardil wrote:
> > Hi, I've managed to get HYLAFAX>T38MODEM->
> > ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk
> > drops a message telling "Unknown RTP codec 96 received from gateway" Do
> > somebody know how to fix it ?
>
> There's nothing to fix; the gateway sent an expected RTP packet, which
> Asterisk dropped. If your FAX works, then you can ignore this.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] Polycom Spectralink 8002 WiFi Phones

[Cesar Gonzalez]

Michael Graves wrote:
> On Tue, 14 Jul 2009 22:58:14 + (UTC), Jeff LaCoursiere wrote:
>
>   
>>> Jeff, yeah i saw the posts, i followed Bob Pierce config and had no
>>> luck, BUT it just started to work, i changed AP's, seems like theres
>>> something wrong with Ubiquiti NanoStation2 WMM implementation, i used a
>>> Linksys WRT54G2 and viola! it started to work, i guess i should've done
>>> that to begin with... :(
>>>
>>> I'll play around whit the Nanostations QoS settings and see if i can get
>>> it to work on those AP's.
>>>
>>> What AP's were you using?
>>>   
>> Hi Cesar,
>>
>> I did actually get it to work as well, and was using Linksys WRT54G with 
>> dd-wrt.  I *intended* for the phone to be useful at random wifi hotspots, 
>> however, and was a bit disappointed to find that that was not going to 
>> work.  So it sits on a shelf gathering dust...
>> 
>
> I had one of these for evalution last spring. The resulting review is
> here:
>
> http://www.smallnetbuilder.com/content/view/30498/80/
>
> They work well enough when paired with suitable APs. According to
> Polycom you must support WMM or all bets are off. In my case I had a
> Netgear WRT-2000.
>
> I had no issues at all with integration with Asterisk. However, I think
> that there truly isn't a dedicated Wifi handset that will satisfy if
> you want to be able to roam the world and make calls from public
> hotspots.
>
> Too many hotspots require a web login before you get access. Thus the
> best devices for this sort of thing seem to be Nokia dual mode phones
> with built-in web browsers.
>
> Michael
> --
> Michael Graves
> mgravesmstvp.com
> http://blog.mgraves.org
> o713-861-4005
> c713-201-1262
> sip:mgra...@mstvp.onsip.com
> skype mjgraves
> fwd 54245
>
>
>
>
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>   

Michael, actually the intended use for this phones is at a customers warehouse 
and its trailer/truck yard, not roaming around public hotspots, so its now 
going to come down to selecting the proper AP's as you suggest, we were looking 
at Naonstation2 for their WDS implemetation, so maybe a set of linksys AP's 
with dd-wrt will do the job.

I've been testing it with this Linksys WRT54G2 since yesterday and works great 
with asterisk. 

Great review btw.

-Cesar



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Re: [asterisk-users] Iphone setup

[James Noble]

Thank you for the heads up.  I will look into both weephone and voipover3g
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[asterisk-users] early-dial SIP 484 "incomplete address", dialplan patterns and international calls

[Vieri]

Hi,

I would like to know if someone can suggest me an efficient way of writing a 
dialplan to match "variable-length" international calls when using SIP clients 
with the "early dial" or 484 feature.

What I usually do for clients that do NOT "early dial" is define something like 
this in my outbound context:

For local calls (they fortunately have a fixed length):
exten => _0Z, ...

For international numbers (variable length:
exten => _000ZX., ...

etc.

However, as you can expect, "early dial" phones will work fine for local calls 
but will fail for international numbers because Asterisk will start dialing out 
as soon as it matches, say, 00044, and ignore the digits sent afterwards.

What would be the most efficient dialplan solution for a mixed client 
environment. Should my "_000ZX." logic have to "wait for digits until timeout"?

Suggestions greatly appreciated.

Vieri




  

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[asterisk-users] iax.conf, IP-based access control

[Philipp Kempgen]

The documentation in
http://svn.digium.com/svn/asterisk/branches/1.4/configs/iax.conf.sample
(and http://svn.digium.com/svn/asterisk/branches/1.6.*/configs/iax.conf.sample)
seems slightly wrong.

---
; ... Limited IP based
; access control is allowed by use of "allow" and "deny" keywords. ...
---

"allow" specifies an allowed codec.
It should read:

---
; ... Limited IP based
; access control is allowed by use of "permit" and "deny" keywords. ...
---

Codecs: disallow/allow
Netmasks: deny/permit

I think this does not justify filing a bug.


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] iax.conf, IP-based access control

[Tilghman Lesher]

On Thursday 16 July 2009 12:26:25 Philipp Kempgen wrote:
> I think this does not justify filing a bug.

No, it does.  Go ahead and file it.

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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Re: [asterisk-users] QoS

[Ira]

At 06:37 AM 7/15/2009, you wrote:
>Ours is just internal, but the concept should be the same.  My boss could
>talk on his phone fine until he cranked up Foxnews feed.  Once the video
>started, he couldn't talk on his phone anymore (bad quality or total loss of
>call).

What I've done here is probably a bit extreme, but we've never had a 
problem of any kind with our VOIP calls. it's a house, no more than 2 
calls at a time on cable internet so it might just be that the 
connection is significantly faster than we ever use. I have a Linksys 
router that has the Asterisk box connected to a port marked High and 
the rest of the house is a second router connected to a port on the 
first router flagged as low or regular. I ran separate Cat5 for the 
phones and the computers. If I knew what I was doing I'd get a Linux 
box with 3 ports and have it do everything.

Ira 


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Re: [asterisk-users] AGI to announce temperature from weather.com XML file

[Jared Smith]

On Thu, 2009-07-16 at 04:49 -0700, Trevor Hammonds wrote:
> I would like to have the ability to have Asterisk announce the temperature
> -- not using TTS -- within the dialplan.  

Chapter 9 of "Asterisk: The Future of Telephony" shows you how to build
an AGI script to do just that.  For a free download, check out
www.asteriskdocs.org.  

There are obviously many other ways to do it.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] iax.conf, IP-based access control

[Philipp Kempgen]

Tilghman Lesher schrieb:
> On Thursday 16 July 2009 12:26:25 Philipp Kempgen wrote:
>> I think this does not justify filing a bug.
> 
> No, it does.  Go ahead and file it.

ok. https://issues.asterisk.org/view.php?id=15518


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
-- 

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Re: [asterisk-users] PRI hunt group

[Gondar Monn]

I managed to do it with a simple forward to the one of our DIDs  The
problem now is that I loose the CID of the original caller ... Is there a
way to forward the call with the original caller ID ?

Thanks!

G.

On Wed, Jul 15, 2009 at 5:38 PM, John Novack
wrote:

>
>
> Steve Totaro wrote:
> > Forwarding a POTS line will not work, it is like a trunk to trunk
> > transfer so it is not "free", so the line stays busy.
> >
> > You need to port that number over to the PRI provider.
> >
> That all depends on the POTS provider.
> Multiple calls from one POTS number CAN be done, but finding anyone
> these days in a business office that can do anything the least bit out
> of the ordinary is near impossible, and one may not be willing to pay
> for it.
> Same with repair - The stock answer - "We checked the line from here, it
> has to be your equipment"  even with an open pair in their (
> unmaintained ) Outside Plant  is about all one can expect.
> Also, not all POTS lines are portable. In the US some ILEC's are exempt
> from the provisions of the 1996  telecom Act.
> All above assume this is in the US.
>
> John Novack
> > On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn  > > wrote:
> >
> > Thank you for your quick answers!
> > @ Brent: rollover is on, I would like to any calls that come on
> > 5551234 to another DID, to be able to receive several calls on the
> > same number
> > @ Don: You are right, I am talking about a specific DID: We have
> > an analog line with busy forward setup @ the telco to forward
> > calls to 5551234 ... But 2 lines are not enough, so I would like
> > to do the same locally: All calls to 555-1234 are forwarded to
> > 555-2345 to free up the line  Does that make sense ?
> > To answer your second question: calls to 555-1234 are alerted on
> > the first channel available channel. All subsequent calls to the
> > DID report busy
> >
> > Again, Thanks for helping me out
> > G.
> > On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly  > > wrote:
> >
> > "Rollover" or "hunting" is generally the default on PRIs. It
> > sounds like
> > Gondar's concern is with a specific DID number (Do multiple
> > calls to other
> > DID numbers work OK?). I'd wonder about a couple things:
> >
> > Are people dialing '5551234' directly, or are calls being
> > forwarded to that
> > number? Some call-forwarding schemes will only forward one
> > call at a time,
> > giving other callers a busy signal.
> >
> > Are you sure that calls to '5551234' aren't being alerted on
> > more than one
> > channel and your Asterisk configuration is presenting the busy?
> >
> > --Don
> >
> > Don Kelly
> >
> > PCF Corp
> > People Come First
> > 651 842-1000
> > 888 Don Kell(y)
> > 651 842-1001 fax
> >
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > 
> > [mailto:asterisk-users-boun...@lists.digium.com
> > ] On Behalf Of
> > Brent Davidson
> > Sent: Wednesday, July 15, 2009 5:37 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] PRI hunt group
> >
> > Gondar Monn wrote:
> > > I am having trouble with a DID on a PRI. If there is a call to
> > > that DID (let say 5551234) , the next calls get a busy
> > signal. How to
> > > I go about sending the call to the next available channel ?
> > > Thanks!
> > >
> > > G.
> > >
> > >
> > If the telco is providing the PRI then you need to tell them
> > you want
> > rollover on the PRI's.  Otherwise, anybody calling across the
> > PSTN to
> > the DID number that is bound to the PRI channel is going to
> > get a busy
> > signal from the telco if that channel is in use.
> >
> > ___
> >
> >
> >
> > --
> > Thanks,
> > Steve Totaro
> > +18887771888 (Toll Free)
> > +12409381212 (Cell)
> > +12024369784 (Skype)
> > 
> >
> > ___
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> >
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>
> --
> Dog is my co-pilot
>
>
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Re: [asterisk-users] setvar and transfer

[Benny Amorsen]

Philipp Kempgen  writes:

> Benny Amorsen schrieb:
>
>> Last concern: Does setvar work even for transfers, like accountcode
>> does?
>
> I can't answer your question, but transfer != transfer. Some use
> a feature code in Asterisk, some initiate a transfer on their phone,
> some use a way to call the Transfer() application.
> Mixing it up causes a lot of confusion.

Sorry, I meant "Does setvar work even for SIP blind transfers during
calls, like accountcode codes?"


/Benny


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Re: [asterisk-users] PRI hunt group

[Don Kelly]

Looks like the caller ID gets lost when you forward. Normally if you do a
"simple forward" using central office features, the caller ID will be the
calling party's number. If you're using a PBX (or something that looks like
one) the PBX does a hook-flash, makes a call to your PRI DID and you see the
number of the line doing the forwarding. As someone pointed out earlier,
port it if you can.

--Don

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gondar Monn
Sent: Thursday, July 16, 2009 1:56 PM
To: novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] PRI hunt group

 

I managed to do it with a simple forward to the one of our DIDs  The
problem now is that I loose the CID of the original caller ... Is there a
way to forward the call with the original caller ID ?

Thanks!

G.

On Wed, Jul 15, 2009 at 5:38 PM, John Novack 
wrote:



Steve Totaro wrote:
> Forwarding a POTS line will not work, it is like a trunk to trunk
> transfer so it is not "free", so the line stays busy.
>
> You need to port that number over to the PRI provider.
>

That all depends on the POTS provider.
Multiple calls from one POTS number CAN be done, but finding anyone
these days in a business office that can do anything the least bit out
of the ordinary is near impossible, and one may not be willing to pay
for it.
Same with repair - The stock answer - "We checked the line from here, it
has to be your equipment"  even with an open pair in their (
unmaintained ) Outside Plant  is about all one can expect.
Also, not all POTS lines are portable. In the US some ILEC's are exempt
from the provisions of the 1996  telecom Act.
All above assume this is in the US.

John Novack

> On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn  > wrote:
>
> Thank you for your quick answers!
> @ Brent: rollover is on, I would like to any calls that come on
> 5551234 to another DID, to be able to receive several calls on the
> same number
> @ Don: You are right, I am talking about a specific DID: We have
> an analog line with busy forward setup @ the telco to forward
> calls to 5551234 ... But 2 lines are not enough, so I would like
> to do the same locally: All calls to 555-1234 are forwarded to
> 555-2345 to free up the line  Does that make sense ?
> To answer your second question: calls to 555-1234 are alerted on
> the first channel available channel. All subsequent calls to the
> DID report busy
>
> Again, Thanks for helping me out
> G.
> On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly  > wrote:
>
> "Rollover" or "hunting" is generally the default on PRIs. It
> sounds like
> Gondar's concern is with a specific DID number (Do multiple
> calls to other
> DID numbers work OK?). I'd wonder about a couple things:
>
> Are people dialing '5551234' directly, or are calls being
> forwarded to that
> number? Some call-forwarding schemes will only forward one
> call at a time,
> giving other callers a busy signal.
>
> Are you sure that calls to '5551234' aren't being alerted on
> more than one
> channel and your Asterisk configuration is presenting the busy?
>
> --Don
>
> Don Kelly
>
> PCF Corp
> People Come First
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> 
> [mailto:asterisk-users-boun...@lists.digium.com
> ] On Behalf Of
> Brent Davidson
> Sent: Wednesday, July 15, 2009 5:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] PRI hunt group
>
> Gondar Monn wrote:
> > I am having trouble with a DID on a PRI. If there is a call to
> > that DID (let say 5551234) , the next calls get a busy
> signal. How to
> > I go about sending the call to the next available channel ?
> > Thanks!
> >
> > G.
> >
> >
> If the telco is providing the PRI then you need to tell them
> you want
> rollover on the PRI's.  Otherwise, anybody calling across the
> PSTN to
> the DID number that is bound to the PRI channel is going to
> get a busy
> signal from the telco if that channel is in use.
>
> ___
>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)

> 

>
> __

[asterisk-users] possible to configure 2 servers - one is backup system for the other?

[Norbert Zawodsky]

Hello everybody!

Please let me ask you a question:

Is it possible (and if yes, how) to configure 2 asterisk servers on two
machines so that the second one acts as a backup system if the first one
is unresponsive?

Clearly, the second should take over automacigally (but not necessarily
during an active call).

Many thanks for your time!

Norbert


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Re: [asterisk-users] possible to configure 2 servers - one is backup system for the other?

[Tim Nelson]

- "Norbert Zawodsky"  wrote:
> Hello everybody!
> 
> Please let me ask you a question:
> 
> Is it possible (and if yes, how) to configure 2 asterisk servers on
> two
> machines so that the second one acts as a backup system if the first
> one
> is unresponsive?
> 
> Clearly, the second should take over automacigally (but not
> necessarily
> during an active call).
> 
> Many thanks for your time!
> 
> Norbert
> 

Check out Linux-HA using heartbeat and a disk replication system like DRBD. 
Solid.

--Tim

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Re: [asterisk-users] setvar and transfer

[Danny Nicholas]

I'm going to give a "qualified" no.  The reason being is that setvar works
in a session (say SIP/100-abcdefg) and the blind transfer may spawn a new
session like Local/1-abcdefg).  So your only solid variables are the global
ones.  You can verify this by looking at CLI output with verbose set to at
least 5. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen
Sent: Thursday, July 16, 2009 2:18 PM
To: Philipp Kempgen
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setvar and transfer

Philipp Kempgen  writes:

> Benny Amorsen schrieb:
>
>> Last concern: Does setvar work even for transfers, like accountcode
>> does?
>
> I can't answer your question, but transfer != transfer. Some use
> a feature code in Asterisk, some initiate a transfer on their phone,
> some use a way to call the Transfer() application.
> Mixing it up causes a lot of confusion.

Sorry, I meant "Does setvar work even for SIP blind transfers during
calls, like accountcode codes?"


/Benny


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Re: [asterisk-users] How to Change size of CDR(accountcode) variable?

[Jared Smith]

On Tue, 2009-07-14 at 00:01 +0200, Benny Amorsen wrote:
> Last concern: Does setvar work even for transfers, like accountcode
> does?

At least in theory, the setvar= setting in sip.conf or iax.conf (or in
Asterisk 1.6.0 and later, chan_dahdi.conf) should work just like the
Set() dialplan application, in that you can prepend an underscore or two
to the variable name to make it inheritable by spawned channels.

So, in theory, "setvar=_FANCYLONGACCOUNTCODE=foo" should make that
channel variable inheritable by the *next* spawned channel (but not any
channels beyond that), and "setvar=__FANCYLONGACCOUNTCODE=foo" should
make it inheritable by the spawned channel *and* any channels it spawns,
and so forth.

That being said, it's just theory.  I have not tested this in my lab,
but I offer it as a simple suggestion for you to try.  Please let us
know if this helped.  (My gut feeling is that it should work for DTMF
and flash-based transfers.  I'm a little less sure about SIP-initiated
transfers.)

-- 
Jared Smith
Training Manager
Digium, Inc.


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[asterisk-users] 800 number portability

[Jeff LaCoursiere]

Apologies for the off topic post... hoping someone knows if 800 number 
portability in the states is legally enforced?  One of my customers is 
being told by their current "vanity" 800 provider that they own the number 
and refuse to release it to their new carrier.  I thought I understood 
that in 1991 the FCC mandated portability by 1993.  Are they bluffing? 
They want a 3 year buyout to release the number!

j

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[asterisk-users] Unique id used for call recording missing from CDR data for transferred call

[Scott Gifford]

Hello,

I have an application that needs to record outgoing calls.  It's
running on Asterisk 1.4.18, with CDR data stored in MySQL.

Outgoing calls are recorded based on their uniqueid.  When outgoing
calls are placed, there is a line like this on my extensions.conf:

exten => _.,n,MixMonitor(/var/spool/asterisk/monitor/${UNIQUEID}.gsm)

For regular outgoing calls, this works fine.  The call is recorded in
a file named for its uniqueid, and if I need the recording I can pull
the information out of the CDR table, find the uniqueid, then pull up
the recording.  We have a Web application that does this
automatically.

When an incoming is transfered to an outgoing call via an attended SIP
transfer, however, the uniqueid assigned to the outgoing call does not
seem to end up in the CDR table at all.

Here's what happens:

  1. User A calls in to the queue on Zap/1-1 uniqueid 1247686911.203
  2. Operator picks up call.
  3. Queue application starts recording in 1247686911.203.gsm
  4. Operator talks for awhile, decides to transfer call
  5. Operator switches Line 2, calls User B on channel Zap/24-1 with
 uniqueid 1247686911.205
  6. Dialplan command MixMonitor(1247686911.205.gsm) starts recording
  7. Operator talks to User B for awhile
  8. Operator transfers call, connects User A to User B, and hangs up

This generates these 4 CDRs:

  1. 2009-07-15 15:41:51
 channel Zap/1-1
 dstchannel SIP/DEV-0078ebd0
 duration 175
 uniqueid 1247686911.203

  2. 2009-07-15 15:43:17
 channel SIP/DEV-0078ebd0
 dstchannel (none)
 duration 89
 uniqueid 1247686997.204

  3. 2009-07-15 15:43:47
 channel Zap/24-1
 dstchannel Zap/1-1
 duration 80
 uniqueid 1247687027.206

  4. 2009-07-15 15:43:47
 channel SIP/DEV-c4018700
 dstchannel Zap/24-1
 duration 59
 uniqueid 1247686911.203

The uniqueid for the outgoing call, 1247686911.205, doesn't end up in
any of the CDRs, so is effectively lost.

Any ideas on how to handle this?  Is this something that's likely to
be fixed in a later version of Asterisk?

Thanks,

Scott.

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[asterisk-users] Stop recording on SIP attended transfer

[Scott Gifford]

Hello,

We have an application where operators will sometimes take an incoming
call from a queue, then contact an outside line, do a consultation,
and finally do a SIP attended transfer to join the two parties
together.  We'd like to record the incoming caller's conversation with
the operator and the attended part of the outgoing call, but not the
unattended part, after the transfer has completed and the incoming
caller is talking on the outgoing line (this part of the call may be
confidential).

We start the recording with a MixMonitor command when the outgoing
call is placed.  However, I don't see anything in the dialplan that
gets run when the SIP attended transfer happens, where I could issue a
command to stop recording.

Any suggestions?

Thanks!

Scott.

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Re: [asterisk-users] Stop recording on SIP attended transfer

[Danny Nicholas]

I don't know the full details, but I think if the Dial command(s) have the W
and/or w options on them, you can activate/deactivate recording via DTMF.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Gifford
Sent: Thursday, July 16, 2009 4:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Stop recording on SIP attended transfer

Hello,

We have an application where operators will sometimes take an incoming
call from a queue, then contact an outside line, do a consultation,
and finally do a SIP attended transfer to join the two parties
together.  We'd like to record the incoming caller's conversation with
the operator and the attended part of the outgoing call, but not the
unattended part, after the transfer has completed and the incoming
caller is talking on the outgoing line (this part of the call may be
confidential).

We start the recording with a MixMonitor command when the outgoing
call is placed.  However, I don't see anything in the dialplan that
gets run when the SIP attended transfer happens, where I could issue a
command to stop recording.

Any suggestions?

Thanks!

Scott.

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Re: [asterisk-users] 800 number portability

[Steve Totaro]

On Thu, Jul 16, 2009 at 5:39 PM, Jeff LaCoursiere  wrote:

>
> Apologies for the off topic post... hoping someone knows if 800 number
> portability in the states is legally enforced?  One of my customers is
> being told by their current "vanity" 800 provider that they own the number
> and refuse to release it to their new carrier.  I thought I understood
> that in 1991 the FCC mandated portability by 1993.  Are they bluffing?
> They want a 3 year buyout to release the number!
>
> j
>

It may depend on the way the "vanity" number was obtained.  I bought mine
from tollfreenumbers.com or whatever and they are mine with docs to back it
up.

Perhaps if obtained in another fashion, the fine print may say differently.

More explanation of where the numbers came from and any contractual
obligations may help.

Similarly, there used to be and probably still are places that would allow
you to register a domain name for free or close to it.  In the fine print
was the fact that the domain name was not yours, they registered it on your
behalf, in their name.

A year later you got a renewal bill or some such and if you wanted to move
it, you had to pay through the nose.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] 800 number portability

[Don Kelly]

Changing toll-free RespOrgs (Responsible Organizations) is different from
number portability. 

That said, the owner of a toll-free number has the right to change RespOrgs,
so the question is "Who is the owner?"

Has your customer been buying simple toll-free service and owned the number
all along, or are they buying some sort of enhanced service and the provider
owns the number?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, July 16, 2009 4:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 800 number portability


Apologies for the off topic post... hoping someone knows if 800 number 
portability in the states is legally enforced?  One of my customers is 
being told by their current "vanity" 800 provider that they own the number 
and refuse to release it to their new carrier.  I thought I understood 
that in 1991 the FCC mandated portability by 1993.  Are they bluffing? 
They want a 3 year buyout to release the number!

j

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Re: [asterisk-users] PRI hunt group

[Steve Totaro]

Try to get a level one tech to set RDNIS on your forwarded POTS line.

Good luck!

On Thu, Jul 16, 2009 at 3:30 PM, Don Kelly  wrote:

>  Looks like the caller ID gets lost when you forward. Normally if you do a
> “simple forward” using central office features, the caller ID will be the
> calling party’s number. If you’re using a PBX (or something that looks like
> one) the PBX does a hook-flash, makes a call to your PRI DID and you see the
> number of the line doing the forwarding. As someone pointed out earlier,
> port it if you can.
>
> --Don
>
>
>  --
> R
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gondar Monn
> *Sent:* Thursday, July 16, 2009 1:56 PM
> *To:* novacks...@gmail.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
>
> *Subject:* Re: [asterisk-users] PRI hunt group
>
>
>
> I managed to do it with a simple forward to the one of our DIDs  The
> problem now is that I loose the CID of the original caller ... Is there a
> way to forward the call with the original caller ID ?
>
> Thanks!
>
> G.
>
> On Wed, Jul 15, 2009 at 5:38 PM, John Novack <
> jnov...@stromberg-carlson.org> wrote:
>
>
>
> Steve Totaro wrote:
> > Forwarding a POTS line will not work, it is like a trunk to trunk
> > transfer so it is not "free", so the line stays busy.
> >
> > You need to port that number over to the PRI provider.
> >
>
> That all depends on the POTS provider.
> Multiple calls from one POTS number CAN be done, but finding anyone
> these days in a business office that can do anything the least bit out
> of the ordinary is near impossible, and one may not be willing to pay
> for it.
> Same with repair - The stock answer - "We checked the line from here, it
> has to be your equipment"  even with an open pair in their (
> unmaintained ) Outside Plant  is about all one can expect.
> Also, not all POTS lines are portable. In the US some ILEC's are exempt
> from the provisions of the 1996  telecom Act.
> All above assume this is in the US.
>
> John Novack
>
> > On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn 
> > > wrote:
> >
> > Thank you for your quick answers!
> > @ Brent: rollover is on, I would like to any calls that come on
> > 5551234 to another DID, to be able to receive several calls on the
> > same number
> > @ Don: You are right, I am talking about a specific DID: We have
> > an analog line with busy forward setup @ the telco to forward
> > calls to 5551234 ... But 2 lines are not enough, so I would like
> > to do the same locally: All calls to 555-1234 are forwarded to
> > 555-2345 to free up the line  Does that make sense ?
> > To answer your second question: calls to 555-1234 are alerted on
> > the first channel available channel. All subsequent calls to the
> > DID report busy
> >
> > Again, Thanks for helping me out
> > G.
> > On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly 
> > > wrote:
> >
> > "Rollover" or "hunting" is generally the default on PRIs. It
> > sounds like
> > Gondar's concern is with a specific DID number (Do multiple
> > calls to other
> > DID numbers work OK?). I'd wonder about a couple things:
> >
> > Are people dialing '5551234' directly, or are calls being
> > forwarded to that
> > number? Some call-forwarding schemes will only forward one
> > call at a time,
> > giving other callers a busy signal.
> >
> > Are you sure that calls to '5551234' aren't being alerted on
> > more than one
> > channel and your Asterisk configuration is presenting the busy?
> >
> > --Don
> >
> > Don Kelly
> >
> > PCF Corp
> > People Come First
> > 651 842-1000
> > 888 Don Kell(y)
> > 651 842-1001 fax
> >
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > 
> > [mailto:asterisk-users-boun...@lists.digium.com
> > ] On Behalf Of
> > Brent Davidson
> > Sent: Wednesday, July 15, 2009 5:37 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] PRI hunt group
> >
> > Gondar Monn wrote:
> > > I am having trouble with a DID on a PRI. If there is a call to
> > > that DID (let say 5551234) , the next calls get a busy
> > signal. How to
> > > I go about sending the call to the next available channel ?
> > > Thanks!
> > >
> > > G.
> > >
> > >
> > If the telco is providing the PRI then you need to tell them
> > you want
> > rollover on the PRI's.  Otherwise, anybody calling across the
> >

[asterisk-users] Compilation error

[michel freiha]

Hi all

I'm trying to install asteris 1.4.22.1 on Solaris 10...the server is V120
SUN spark...During compilation (gmake) I got the following error

/vis.c -o np/vis.o_a
np/vis.c: In function `svis':
np/vis.c:205: error: `u_int32_t' undeclared (first use in this function)
np/vis.c:205: error: (Each undeclared identifier is reported only once
np/vis.c:205: error: for each function it appears in.)
np/vis.c:205: error: syntax error before "u_char"
np/vis.c:205: error: syntax error before ')' token
np/vis.c:205: error: syntax error before "u_char"
np/vis.c:205: error: syntax error before ')' token
np/vis.c:207: error: syntax error before "u_char"
np/vis.c:207: error: syntax error before ')' token
np/vis.c:207: error: syntax error before "u_char"
np/vis.c:207: error: syntax error before ')' token
np/vis.c: In function `strsvis':
np/vis.c:245: error: `u_int32_t' undeclared (first use in this function)
np/vis.c:245: error: syntax error before "u_char"
np/vis.c:245: error: syntax error before ')' token
np/vis.c:245: error: syntax error before "u_char"
np/vis.c:245: error: syntax error before ')' token
np/vis.c:248: error: syntax error before "u_char"
np/vis.c:248: error: syntax error before ')' token
np/vis.c:248: error: syntax error before "u_char"
np/vis.c:248: error: syntax error before ')' token
np/vis.c: In function `strsvisx':
np/vis.c:275: error: `u_int32_t' undeclared (first use in this function)
np/vis.c:275: error: syntax error before "u_char"
np/vis.c:275: error: syntax error before ')' token
np/vis.c:275: error: syntax error before "u_char"
np/vis.c:275: error: syntax error before ')' token
np/vis.c:280: error: syntax error before "u_char"
np/vis.c:280: error: syntax error before ')' token
np/vis.c:280: error: syntax error before "u_char"
np/vis.c:280: error: syntax error before ')' token
np/vis.c: In function `vis':
np/vis.c:303: error: `u_int32_t' undeclared (first use in this function)
np/vis.c:303: error: syntax error before "u_char"
np/vis.c:303: error: syntax error before ')' token
np/vis.c:303: error: syntax error before "u_char"
np/vis.c:303: error: syntax error before ')' token
np/vis.c:305: error: syntax error before "u_char"
np/vis.c:305: error: syntax error before ')' token
np/vis.c:305: error: syntax error before "u_char"
np/vis.c:305: error: syntax error before ')' token
gmake[2]: *** [np/vis.o_a] Error 1
gmake[1]: *** [editline/libedit.a] Error 2
gmake: *** [main] Error 2

I guess there is a BUG in vis.c...Can you please let me know how to define
this variable?

Thanks
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Re: [asterisk-users] 800 number portability

[Jeff LaCoursiere]

On Thu, 16 Jul 2009, Don Kelly wrote:

> Changing toll-free RespOrgs (Responsible Organizations) is different from
> number portability.
>
> That said, the owner of a toll-free number has the right to change RespOrgs,
> so the question is "Who is the owner?"

The "owner" in this case is "CallSource" (www.callsource.com).  Funny 
enough, it looks a lot like the kind of stuff you do, Don ;)

So I guess my disconnect is that a party can "own" an 800 number, but have 
it routed by the RespOrg of their choice?  In this case my client must be 
renting the 800 number from CallSource, and they are the actual owner, so 
are refusing to let it go.  Does that sound right?

>
> Has your customer been buying simple toll-free service and owned the number
> all along, or are they buying some sort of enhanced service and the provider
> owns the number?

I assumed it was simple 800 service (and in fact at first they told me it 
was AT&T they were getting the service from).  It seems that this is 
actually something enhanced.

Cheers,

j

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Re: [asterisk-users] 800 number portability

[Cary Fitch]

There are national number "rental agencies" that lease out prime 800 numbers
even down to the rate center level.

They own the number, not the "renter", and there is a contract that says so.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, July 16, 2009 7:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 800 number portability


On Thu, 16 Jul 2009, Don Kelly wrote:

> Changing toll-free RespOrgs (Responsible Organizations) is different from
> number portability.
>
> That said, the owner of a toll-free number has the right to change
RespOrgs,
> so the question is "Who is the owner?"

The "owner" in this case is "CallSource" (www.callsource.com).  Funny 
enough, it looks a lot like the kind of stuff you do, Don ;)

So I guess my disconnect is that a party can "own" an 800 number, but have 
it routed by the RespOrg of their choice?  In this case my client must be 
renting the 800 number from CallSource, and they are the actual owner, so 
are refusing to let it go.  Does that sound right?

>
> Has your customer been buying simple toll-free service and owned the
number
> all along, or are they buying some sort of enhanced service and the
provider
> owns the number?

I assumed it was simple 800 service (and in fact at first they told me it 
was AT&T they were getting the service from).  It seems that this is 
actually something enhanced.

Cheers,

j

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Re: [asterisk-users] Stop recording on SIP attended transfer

[Scott Gifford]

"Danny Nicholas"  writes:

> I don't know the full details, but I think if the Dial command(s) have the W
> and/or w options on them, you can activate/deactivate recording via DTMF.

Thanks, that's a good idea, I might be able to rig something up with
that!

Scott.

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[asterisk-users] 2 Problems with 1.6.2

[Ira]

I've been using 1.6.2 for a few weeks and I've managed to get almost 
everything working perfectly.

I can't get the MWI indicators on my Aastra phones to work properly, 
the did in all the versions of 1.2 I used up to the most recent one, 
but now they work correctly right after the phone is re-started and 
rarely thereafter. it's as if something changed in the way the MWI is 
handled and I can't figure out what the difference is or what I've done wrong.

When I restart the machines I can't make an outgoing DAHDI call until 
I get an incoming call on that same line.


Ira


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Re: [asterisk-users] PRI hunt group

[C F]

In the good old days telcos didn't care how many channels your forward
used up, they just did it. However nowadays they only allow one
channel at a time to be forwarded, if you need more you have to pay
for it.
Verizon here in NJ charges around $8.00 a month for each call path
(channel), and so do most CLECs.
Optimum cable used to not restrict how many channels, however the last
customer I tried this with only one channel worked, even though for
the ones I set it up before they started blocking it it still works.

If you don't want to port it to the PRI for whatever reason you can
convert it to a RCFW (remote call forwarded number) which is around
$15.00 plus $8.00 for each additional channel again pricing is for
here in Verizon land.


On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn wrote:
> Thank you for your quick answers!
> @ Brent: rollover is on, I would like to any calls that come on 5551234 to
> another DID, to be able to receive several calls on the same number
> @ Don: You are right, I am talking about a specific DID: We have an analog
> line with busy forward setup @ the telco to forward calls to 5551234 ... But
> 2 lines are not enough, so I would like to do the same locally: All calls to
> 555-1234 are forwarded to 555-2345 to free up the line  Does that make
> sense ?
> To answer your second question: calls to 555-1234 are alerted on the first
> channel available channel. All subsequent calls to the DID report busy
>
> Again, Thanks for helping me out
> G.
> On Wed, Jul 15, 2009 at 4:21 PM, Don Kelly  wrote:
>>
>> "Rollover" or "hunting" is generally the default on PRIs. It sounds like
>> Gondar's concern is with a specific DID number (Do multiple calls to other
>> DID numbers work OK?). I'd wonder about a couple things:
>>
>> Are people dialing '5551234' directly, or are calls being forwarded to
>> that
>> number? Some call-forwarding schemes will only forward one call at a time,
>> giving other callers a busy signal.
>>
>> Are you sure that calls to '5551234' aren't being alerted on more than one
>> channel and your Asterisk configuration is presenting the busy?
>>
>> --Don
>>
>> Don Kelly
>>
>> PCF Corp
>> People Come First
>> 651 842-1000
>> 888 Don Kell(y)
>> 651 842-1001 fax
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent
>> Davidson
>> Sent: Wednesday, July 15, 2009 5:37 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] PRI hunt group
>>
>> Gondar Monn wrote:
>> > I am having trouble with a DID on a PRI. If there is a call to
>> > that DID (let say 5551234) , the next calls get a busy signal. How to
>> > I go about sending the call to the next available channel ?
>> > Thanks!
>> >
>> > G.
>> >
>> >
>> If the telco is providing the PRI then you need to tell them you want
>> rollover on the PRI's.  Otherwise, anybody calling across the PSTN to
>> the DID number that is bound to the PRI channel is going to get a busy
>> signal from the telco if that channel is in use.
>>
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>
>
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Re: [asterisk-users] 2 Problems with 1.6.2

[Tilghman Lesher]

On Thursday 16 July 2009 20:19:07 Ira wrote:
> I've been using 1.6.2 for a few weeks and I've managed to get almost
> everything working perfectly.
>
> I can't get the MWI indicators on my Aastra phones to work properly,
> the did in all the versions of 1.2 I used up to the most recent one,
> but now they work correctly right after the phone is re-started and
> rarely thereafter. it's as if something changed in the way the MWI is
> handled and I can't figure out what the difference is or what I've done
> wrong.
>
> When I restart the machines I can't make an outgoing DAHDI call until
> I get an incoming call on that same line.

It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt,
and UPGRADE.txt, as the issue with MWI is explained in there and what you can
do to "fix" it.  The second file contains the explanation, although you would
be well advised to read all three.

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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Re: [asterisk-users] 800 number portability

[Don Kelly]

On Thu, 16 Jul 2009, Don Kelly wrote:

> Changing toll-free RespOrgs (Responsible Organizations) is different from
> number portability.
>
> That said, the owner of a toll-free number has the right to change
RespOrgs,
> so the question is "Who is the owner?"

The "owner" in this case is "CallSource" (www.callsource.com).  Funny 
enough, it looks a lot like the kind of stuff you do, Don ;)

So I guess my disconnect is that a party can "own" an 800 number, but have 
it routed by the RespOrg of their choice?  In this case my client must be 
renting the 800 number from CallSource, and they are the actual owner, so 
are refusing to let it go.  Does that sound right?

>
> Has your customer been buying simple toll-free service and owned the
number
> all along, or are they buying some sort of enhanced service and the
provider
> owns the number?

I assumed it was simple 800 service (and in fact at first they told me it 
was AT&T they were getting the service from).  It seems that this is 
actually something enhanced.

Cheers,

j

Yes, someone can "own" a number and transfer service from one RespOrg to
another to save a few bucks or get better or different service. If your
client originally got the number from CallSource, it probably belongs to
CallSource. If they got it from someone else and transferred to CallSource
they would have a different argument.

At www.callsource.com/services/numbers.php CallSource says "Own your
tracking numbers. CallSource is one of the few call tracking vendors that
will transfer permanent ownership of numbers to you." I think the reasonable
person would rely on this statement and not expect they would want
three-years' payment for it. Does your customer have their original
contract?

  --Don

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Re: [asterisk-users] Mexican ITSP needed

[Michiel van Baak]

On 11:39, Thu 16 Jul 09, Carlos Chavez wrote:
>   Try http://www.inext.com.mx they can provide DIDs in several cities in
> Mexico.

Thanks.
I asked the customer to have a look (I'm only capable of reading English
and Dutch ;))

You have any experience with them ?

> 
> On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote:
> > Hey all,
> > 
> > I was wondering if anyone knows about a Mexican ITSP I can connect to to
> > route calls from and to my * boxen.
> > 
> > If it matters: I'm located in The Netherlands and one of our customers
> > is in Mexico so if we need a Mexican presence that is not an issue.
> > 
> > Thanks.
> > 
-- 

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

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


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Re: [asterisk-users] PRI hunt group

[Alex Balashov]

C F wrote:

> If you don't want to port it to the PRI for whatever reason you can
> convert it to a RCFW (remote call forwarded number) which is around
> $15.00 plus $8.00 for each additional channel again pricing is for
> here in Verizon land.

Is that true even if the number is out of a rate center that is billed 
long-distance relative to the destination (but still intra-LATA)?  Or do 
you pay normal LD rates on top of all that in the intra-LATA LD scenario?

-- 
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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[asterisk-users] Asterisk Error

[michel freiha]

Hi all,

Can you please let me know what the below issue mean when trying to start
asterisk and how I can fix it?

pbx_dundi.c: No ethernet interface found for seeding global EID  You will
have to set it manually.

regards
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[asterisk-users] MoH - can the volume be adjusted

[Rupert Utteridge - Digital Techniques (Austalia) Limited]

We have noticed MoH volume levels vary, very much depending on the terminal
device that is connected. Within Asterisk is there any AGC or level control
available to compensate for the varying terminal devices and their levels?

 

For example a Polycom IP 7000 has very audible level while X Lite on a Dell
laptop has very low level and Cisco 7970G have lower level than the Polycom
IP 7000.

 

What is the experience of other users and how have you handled this level
variation?

 

Rupert

 

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[asterisk-users] [HELP] - Conference bridge

[Rupert Utteridge - Digital Techniques (Austalia) Limited]

Currently the conference bridge in Asterisk can be set for a maximum of 99
hours. For normal use this is more than adequate. However, we have a
requirement to have the conference bridge permanently set up with no maximum
time.

 

Does anyone have experience on the possibility of changing this setting.

 

Rupert 

 

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[asterisk-users] Skill based routing

[Rupert Utteridge - Digital Techniques (Austalia) Limited]

We are trying to implement skill based routing for agents in a support
centre based on the agent login. Has anyone had any experience with this and
what was the outcome?

 

Can anyone share their ideas on this?

 

Rupert

 

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Re: [asterisk-users] [HELP] - Conference bridge

[Alex Balashov]

Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:

> Currently the conference bridge in Asterisk can be set for a maximum of 
> 99 hours. For normal use this is more than adequate. However, we have a 
> requirement to have the conference bridge permanently set up with no 
> maximum time.

Where did you hear of this limitation?


Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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