Re: [asterisk-users] zapata.conf

[Tzafrir Cohen]

On Mon, Apr 09, 2007 at 10:43:27PM -0700, [EMAIL PROTECTED] wrote:
> I have a Digium TDM400b11, 1FXO [port2] & 1FXS [port 1]
> 
> When I reload the chan_zap I get:
> 
>  [chan_zap.so] => (Zapata Telephony w/PRI)
>   == Parsing '/etc/asterisk/zapata.conf': Found
> Apr  9 22:39:36 ERROR[3541]: chan_zap.c:10388 setup_zap: Signalling must be 
> specified before any channels are.
> Apr  9 22:39:36 WARNING[3541]: loader.c:414 __load_resource: chan_zap.so: 
> load_module failed, returning -1
> Apr  9 22:39:36 WARNING[3541]: loader.c:554 load_modules: Loading module 
> chan_zap.so failed!
> 
> Here is my zapata.conf
> cat /etc/asterisk/zapata.conf
> [channels]
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echotraining=yes
> immediate=no
> ;
> ;===
> ; define the channels
> signaling=fxo_ks

signalling=fxo_ks

(note the double 'l')

> context=internal
> channel => 1
> 
> signaling=fxs_ks
> context=incoming
> channel => 2

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Strange error, "logger.c: No more room in scheduler..."

[Massimo Nuvoli]

I found no info about this strange error:

"logger.c: No more room in scheduler"
"logger.c: Asked to delete sched id -1???"

Only in "verbose" mode. Someone know how to solve this?

Asterisk 1.2.13 with sangoma A104EC

Hints?

Thnks.
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email;internet:[EMAIL PROTECTED]
title:Amministratore Delegato
tel;work:0121303544
tel;fax:0121040601
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[asterisk-users] zapata.conf

[ctotos]

I have a Digium TDM400b11, 1FXO [port2] & 1FXS [port 1]

When I reload the chan_zap I get:

 [chan_zap.so] => (Zapata Telephony w/PRI)
  == Parsing '/etc/asterisk/zapata.conf': Found
Apr  9 22:39:36 ERROR[3541]: chan_zap.c:10388 setup_zap: Signalling must be 
specified before any channels are.
Apr  9 22:39:36 WARNING[3541]: loader.c:414 __load_resource: chan_zap.so: 
load_module failed, returning -1
Apr  9 22:39:36 WARNING[3541]: loader.c:554 load_modules: Loading module 
chan_zap.so failed!

Here is my zapata.conf
cat /etc/asterisk/zapata.conf
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;
;===
; define the channels
signaling=fxo_ks
context=internal
channel => 1

signaling=fxs_ks
context=incoming
channel => 2

and here is my zaptel
zaptel.conf
[EMAIL PROTECTED] ~]# grep -v "^#" /etc/zaptel.conf
loadzone = us
defaultzone=us
fxsks=2
fxoks=1

Any suggestion?
-- 
Thanks
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[asterisk-users] Call forwarding (from PHONE configuration) with PRI

[Barry D. Hassler]

Hi folks.

My client is wanting to use call forwarding configured on their phones
(Linksys SPA942), with a PRI from their provider. When we configure call
forwarding, we invariably get a "The number you have dialed is not in
service" message from the providers.

Examining the detailed dial plan debugging as well as the PRI debugging,
the number is dialed correctly. The only difference noticed between a
forwarded call versus a "normal" outbound call is that an extended
attribute of "Forwarded Unconditionally".

The (slightly edited -- I've edited the phone numbers to protect the
innocent) trace is below. Is this an asterisk issue, or an ILEC issue? I
have a ticket open with the ILEC as well, but they tend to be less than
helpful (their response has been to turn on their call forwarding option).

Any thoughts appreciated!


[ 02 01 01 1e ]

   -- Got SIP response 302 "Moved Temporarily" back from 192.168.1.55
mail*CLI>

   -- Now forwarding Zap/1-1 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/2105-08217980)
   -- Executing Macro("Local/[EMAIL PROTECTED],2",
"to-pstn-standard|1937000") in new stack
   -- Executing NoOp("Local/[EMAIL PROTECTED],2",
""CALLERID(num)=" 937001") in new stack
   -- Executing NoOp("Local/[EMAIL PROTECTED],2",
""MACRO_EXTEN=" 1937000") in new stack
   -- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?:7") in new stack
   -- Goto (macro-to-pstn-standard,s,7)
   -- Executing NoOp("Local/[EMAIL PROTECTED],2", ""Call
Forwarded"") in new stack
   -- Executing Set("Local/[EMAIL PROTECTED],2",
"CALLERID(name)="CLIENT NAME"") in new stack
   -- Executing Set("Local/[EMAIL PROTECTED],2",
"CALLERID(num)=513001") in new stack
   -- Executing Dial("Local/[EMAIL PROTECTED],2",
"Zap/g1/1937000||r") in new stack
   -- Requested transfer capability: 0x00 - SPEECH
mail*CLI>


[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer

capability: Speech (0)

 Ext: 1  Trans mode/rate: 64kbps,

circuit-mode (16)

 Ext: 1  User information layer 1: u-Law

(34)

[18 03 a9 83 82]
Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive

Dchan: 0

   ChanSel: Reserved
  Ext: 1  Coding: 0   Number Specified   Channel

Type: 3

  Ext: 1  Channel: 2 ]
[1e 02 80 83]
Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)

0: 0   Location: User (0)

  Ext: 1  Progress Description: Calling

equipment is non-ISDN. (3) ]

[28 19 b1 22 46 57 44 20 46 52 4f 4d 20 42 4c 55 45 20 4c 4f 4f 50 20

4c 4c 43 22]

Display (len=25) Charset: 31 [ "FWD FROM x" ]
[6c 0c 49 80 35 31 33 32 30 34 32 31 30 30]
Calling Number (len=14) [ Ext: 0  TON: Subscriber Number (4)  NPI:

Private Numbering Plan (9)

  Presentation: Presentation permitted, user

number not screened (0) '513001' ]

[70 0c c9 31 39 33 37 34 32 37 39 30 30 30]
Called Number (len=14) [ Ext: 1  TON: Subscriber Number (4)  NPI:

Private Numbering Plan (9) '1937000' ]

[74 07 49 01 8f 32 31 30 35]
Redirecting Number (len= 9) [ Ext: 0  TON: Subscriber Number (4)  NPI:

Private Numbering Plan (9)

  Ext: 0 Presentation: Presentation

permitted, user number passed network screening (1)

  Ext: 1 Reason: Forwarded unconditionally

(15) '2105' ]
   -- Called g1/1937000
   -- Local/[EMAIL PROTECTED],1 is ringing
mail*CLI>


--
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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Re: [asterisk-users] Cisco GW, PRI & CallerID Name

[Yehavi Bourvine +972-8-9489444]

> Does anybody have callerid name coming in on a Cisco PRI via a Cisco
> gateway via SIP to *?  I've seen a few people ask and a few people that
> say it should work, but I've never seen an actual working config.

I have it working, but it depends on the specific configuration. I have it
working via PRI/Q.sig to a Nortel TX-1. The key issue here is that the name can
be sent in two ways: via the Q.931 and via Q.sig. Nortel supports only Q.sig,
and you need to enable it on the Cisco. If this is your case then tell me and
I'll send the config.

 __Yehavi:
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Re: [asterisk-users] Cisco GW, PRI & CallerID Name

[Alex Balashov]

I have a PRI fed into a Cisco AS5300 media gateway and sent to Asterisk via 
SIP, and caller ID works fine.  I can probably help you figure it out, even 
though I don't have any immediate insights.  Feel free to e-mail me off 
list if you like.


In general, however, I don't have "isdn supp-service name calling" enabled, 
and have my signaling on the voip dial-peer set to "signaling forward 
conditional."


-- Alex

On Mon, 9 Apr 2007, Peder @ NetworkOblivion said something to this effect:

Does anybody have callerid name coming in on a Cisco PRI via a Cisco gateway 
via SIP to *?  I've seen a few people ask and a few people that say it should 
work, but I've never seen an actual working config.


I do a debug on our Cisco gateway and I can see the callerid name, however 
none of the features that should send it via SIP seem to work. Cisco docs say 
to use the following:


voice service voip
signaling forward unconditional

interface serial 1/0:23
isdn supp-service name calling


When I enable either of those features, my calls hangup after about 30 
seconds.  * gives me a message "[Apr  9 22:52:22] WARNING[14660]: 
chan_sip.c:1916 retrans_pkt: Hanging up call 
[EMAIL PROTECTED] - no reply to our critical 
packet."


Turning them off makes the calls work fine.  "signaling forward 
unconditional" appears to be the key feature, but * doesn't seem to know what 
to do with the info that it is sending.  There must be some way to set * to 
decode it, but I can't figure it out.


I am running * 1.4.2 (and I've tried it on 1.0.3 and 1.2.10) and 12.4.13a on 
my Cisco gateway.


Any ideas?  FYI, my gateway has been running fine with multiple * boxes for 
2+ years.  I've finally decided to try and get this working, so I upgraded to 
12.4.13a to see if it worked there and it still doesn't.



Peder

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--
Alex Balashov <[EMAIL PROTECTED]>
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[asterisk-users] Cisco GW, PRI & CallerID Name

[Peder @ NetworkOblivion]

Does anybody have callerid name coming in on a Cisco PRI via a Cisco 
gateway via SIP to *?  I've seen a few people ask and a few people that 
say it should work, but I've never seen an actual working config.


I do a debug on our Cisco gateway and I can see the callerid name, 
however none of the features that should send it via SIP seem to work. 
Cisco docs say to use the following:


voice service voip
 signaling forward unconditional

interface serial 1/0:23
 isdn supp-service name calling


When I enable either of those features, my calls hangup after about 30 
seconds.  * gives me a message "[Apr  9 22:52:22] WARNING[14660]: 
chan_sip.c:1916 retrans_pkt: Hanging up call 
[EMAIL PROTECTED] - no reply to our 
critical packet."


Turning them off makes the calls work fine.  "signaling forward 
unconditional" appears to be the key feature, but * doesn't seem to know 
what to do with the info that it is sending.  There must be some way to 
set * to decode it, but I can't figure it out.


I am running * 1.4.2 (and I've tried it on 1.0.3 and 1.2.10) and 
12.4.13a on my Cisco gateway.


Any ideas?  FYI, my gateway has been running fine with multiple * boxes 
for 2+ years.  I've finally decided to try and get this working, so I 
upgraded to 12.4.13a to see if it worked there and it still doesn't.



Peder

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Re: [asterisk-users] Vonage fraud controls

[Kenneth Padgett]

Has anyone tried pushing calls to a Vonage ATA attached to an FXO card in
Asterisk and had your account terminated by Vonage?

I'm curious as to whether they will stop your service if you push too many
calls through their ATA in a specific period of time.

Thanks in advance for the info, SG


I knew a guy that used two Vonage lines with fax modems attached to
them and pumped "Free" internet email to Fax services through them. As
much as 2000 calls a week per line. It took Vonage many months to
catch on, and they figured out it was a fax service (presumably by
googling or calling the numbers being dialed) . All Vonage did was
force him to upgrade to "Unlimited Business" service on both lines.
They where nasty about it via email, such as "mass faxing is against
our TOS" etc, but they didn't cut his service. Then again, this was a
couple years ago, they may be tougher on it now that they have more
subscribers.

On a side note, faxing over Vonage sucks and is unreliable. Nice voice
service as far as I'm concerned though.

-Kenneth
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Re: [asterisk-users] Verizon-Vonage Lawsuit

[Kenneth Padgett]

"The jury found that three of five disputed patents were infringed and
all are valid, while rejecting Verizon's claim that the infringement was
willful. The patents cover a method of translating calls between the
Internet and standard phones, call-waiting features and wireless handsets."


Is it just me, or does this whole Verizon vs. Vonage seem strikingly
similar to SCO vs. Linux? Where's the supporting documentation?
Where's the patents? I think it's all hush hush because Verizon
believes they can scare people back to them and tarnish Vonage's name,
thus eliminating the biggest competitor they've had in a long time.

I'm not doubting that patents exist, I'm just betting that you'd have
to have some seriously drunken vision to interpret them as the exact
business processes Vonage uses. I think if Verizon thought for a
second they had solid ground to stand on, they would disclose which
patents they're referencing so the public could decide.

I dumped Verizon for Vonage years ago before they have millions of
subscribers. I had more trouble with Verizon than I ever have with
Vonage. All the call quality problems I ever had where easily resolved
with some simple QoS configs on my router. If Vonage goes away, I just
won't have home phone service. I'll never go back to Verizon.

I guess we're all screwed when we find out Vonage runs on Asterisk and
Digium cards!

-Kenneth
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Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 35

[Steve Edwards]

On Mon, 9 Apr 2007, Steve Edwards wrote:


On Tue, 10 Apr 2007, Faisal Ashraf wrote:


I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's
recordings
should go to next date folder.


Using the STRFTIME function to build your file name should get you 

close.


Reading the documentation should fill in the blanks :)


On Tue, 10 Apr 2007, Faisal Ashraf wrote:


We i have settup it like this it giveme caller id agent id and date-time on
gsm file but i want them to be in folder on every day basis datewise.

exten =>
_1NXXNXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP})
exten => _1NXXNXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)

Any Idea ?


Another clue in the same day?

) Build a variable using STRFTIME with a format string of your choice for 
MMDD.


) Call system() passing "mkdir --parents", your recording path and your 
YYMMDD variable. "man mkdir" to refresh your memory on what the command is 
doing :)


) Call monitor like you do above, but insert your YYMMDD variable and a 
couple of slashes and its "Miller time."


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] ${QUEUESTATUS}

[BJ Weschke]

On 4/9/07, Damon Estep <[EMAIL PROTECTED]> wrote:





There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when they
would be set;



If someone could correct errors with these definitions ot would be
appreciated;



TIMEOUT – the max time specified in the queue command elapsed, only checked
between retries so may not be 100% accurate.

FULL – the number of callers in the queues would exceed the maxlen= value
defined in queues.conf if another caller was added

JOINEMPTY – a call was sent to the queue but the queue had no members, does
not apply when using agentcallbacklogin since there could be unavailable
members defined but not available.

LEAVEEMPTY – the last agent was removed form the queue before alls calls we
handled, remong callers exit with this status, also acts differently when
there are only queue members that are unavaialbe

JOINUNAVAIL/LEAVEUNAVAIL – same as JOINEMPTY/LEAVEEMPTY, except that there
were still queue members, but all were status unavailable (logged out)



So if a queue is made up of only callback agents (agentcallbacklogin) then
the queuestatus will never be joinempty or leaveempty



If the maxlen=0 then there will never be a queuestatus of full

If there is no timeout in the queue command thee wll never be a questatus of
timeout

If there are no callback or static agents joinunavail/leaveunavail will
never apply.




I don't believe that the logic you're describing here with "if a
queue is made up of only callback agents then the queuestatus will
never be joinempty or leavempty" is right. If a queue is made up of
only callback agents and none of those callback agents are presently
logged in, the status could certainly be joinempty or leavempty.

The rest of it looks pretty good. Thanks for taking the time to
better document this!

 BJ

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 35

[Faisal Ashraf]

We i have settup it like this it giveme caller id agent id and date-time on
gsm file but i want them to be in folder on every day basis datewise.


exten =>
_1NXXNXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP})
exten => _1NXXNXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)


Any Idea ?

Faisal






--

Message: 16
Date: Tue, 10 Apr 2007 03:08:57 +0500
From: "Faisal Ashraf" <[EMAIL PROTECTED]>
Subject: [asterisk-users] Date Wise Recordings
To: asterisk-users@lists.digium.com
Message-ID:
   <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's
recordings
should go to next date folder.

so how can i do that my current monitor context is like...

exten =>
_1NXXNXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)




Faisal
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--

Message: 19
Date: Mon, 9 Apr 2007 16:01:46 -0700 (PDT)
From: Steve Edwards <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Date Wise Recordings
To: Asterisk Users Mailing List - Non-Commercial Discussion
   
Message-ID: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

On Tue, 10 Apr 2007, Faisal Ashraf wrote:

> I like my recordings to go to date wise folder i mean to say that for
> example today is 20070409 so all recordings should go directly to that
> folder instead of one folder for whole month. and then next day's
recordings
> should go to next date folder.

Using the STRFTIME function to build your file name should get you close.

Reading the documentation should fill in the blanks :)

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

[Rob Hillis]

In the case of our Sangoma card, the echo cancellation module 
constituted approximately half the price of the card, so yes you should 
find it considerably cheaper than the Digium card.


Just be aware of the extra fiddling around having to install the Sangoma 
drivers in addition to the Zaptel drivers.



Bobby Crawford wrote:

The higher price on the Sangoma is for hardware echo cancellation.  There
should be a model (A20400) that doesn't have the echo cancellation and it
probably is less expensive than the Digium card.

Bobby

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze
Sent: Monday, April 09, 2007 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

On Apr 9, 2007, at 9:29 AM, William Moore wrote:

  

On 4/9/07, Jim Freeze <[EMAIL PROTECTED]> wrote:

Or a new Digium TDM880B replacing the old TDM40B for only one  


IRQ...
Do you know if this board will fit in a 2U machine?
  

The TDM800P is about the same height as the TDM400P and is about an
inch longer, so you should have no problem putting it in the same slot
as the TDM400 was in.



I really appreciate all the info. I think I am going to go the route  
of an 8 FXS card,

but still need to decide between Digium and Sangoma.

Are there any recommendations between:

  TDM880B - $679
  A20400D  - $906

Also, why the higher price on the Sangoma?


Jim

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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 35

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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[asterisk-users] no reply to our critical packet

[Joao Pereira]

Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 20 
seconds.
This only happens because Im using Asterisk2Billing's AGI (without 
A2Billing it doesnt hang up).

does someone knows whats the problem??

Here is my Asterisk debug:
(xxx.xxx.xxx.xxx  -> the phone's IP)



Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: 
Unable to spawn mp3player
Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise 
support incomplete in Asterisk (RFC 3389). Please turn off on client if 
possible. Client IP: xxx.xxx.xxx.xxx
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum 
retries exceeded on transmission 
[EMAIL PROTECTED] for seqno 12282 
(Critical Response)
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up 
call [EMAIL PROTECTED] - no reply to 
our critical packet.
Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise 
support incomplete in Asterisk (RFC 3389). Please turn off on client if 
possible. Client IP: xxx.xxx.xxx.xxx



Thanks for the help
Regards
Joao Pereira

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Re: [asterisk-users] Privacy Manager w/ No Recording

[Dovid B]

Is there a way to use privacy manager without requiring the user to enter 
their name?  Essentially I am just looking for a way to force the called 
user to enter "1" to accept the call.  I don't need a name recording.  I 
want a call to come in, a message to be played, music on hold, call out to 
the called party, then enter "1" to accept, "2" to reject.




I posted something to the list a few days ago (I can't fine it in my sent 
emails). You can use what I posted and just cut out the part where it 
records the file as well as where it plays it to the called person.



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[asterisk-users] Re: incoming zaptel calls fail

[Robert La Ferla]

On Apr 9, 2007, at 1:51 PM, "Kevin P. Fleming" <[EMAIL PROTECTED]>  
wrote:

You also neglected to mention the version of Asterisk you are running;
'latest SVN' means nothing when there are 20+ branches of Asterisk on
our SVN server.



Sorry about that.  It is the 1.4 trunk:

Asterisk SVN-branch-1.4-r60850, Copyright (C) 1999 - 2006 Digium,  
Inc. and others.


and to recap:

OS:

Fedora Core 5:

Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686
i686 i386 GNU/Linux

ZAPTEL:

Zaptel Version: SVN-branch-1.4-r2397M


ERROR MSGS:

[Apr  9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID
returned with error on channel 'Zap/2-1'
[Apr  9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID
returned with error on channel 'Zap/2-1'
[Apr  9 09:39:34] WARNING[16580]: chan_zap.c:6470 ss_thread: CallerID
returned with error on channel 'Zap/2-1'

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Re: [asterisk-users] Too much silence, perceived delay

[Stephen Bosch]

Joe Acquisto wrote:
> Stephen Bosch <[EMAIL PROTECTED]> Wrote: 4/9/2007 7:12 PM:
>> Have you been able to test this yourself? (Three to four seconds seems
>> inordinately long. That's as bad as a satellite link.)
> 
> No, not tested by me, I only heard about it today, via email.  

I don't doubt that they are noticing some delay, I just question how
extreme it is.

>> Have you tried tinkering with the gain settings? Adjusting the gain can
>> impact sidetone, which might improve the call experience.
> 
> No, not yet.  Any suggestions as to direction and magnitude?

After confirming that they're experiencing what they say they've been
experiencing, I would start with the rxgain and increment it by 2 or 3,
then test.

(I'd be curious to hear what happens if you turn echo cancellation off.)

-Stephen-
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Re: [asterisk-users] Too much silence, perceived delay

[Joe Acquisto]

Stephen Bosch <[EMAIL PROTECTED]> Wrote: 4/9/2007 7:12 PM:
> Joe Acquisto wrote:
>> In a system connected to a verizon T1, Digium TE411P (quad T1 echo
>> cancellation), client is complaining it is "too quiet".
>> 
>> The complaint regards calls over the T1, not in house SIP only calls.
>> 
>> Their description indicates they want some earpiece feedback of
>> themselves speaking.  Also, they complain that it takes several
>> seconds (3-4) for the other party to respond.  That is kind of
>> subjective, I guess.
> 
> Have you been able to test this yourself? (Three to four seconds seems
> inordinately long. That's as bad as a satellite link.)

No, not tested by me, I only heard about it today, via email.  

> 
> Have you tried tinkering with the gain settings? Adjusting the gain can
> impact sidetone, which might improve the call experience.

No, not yet.  Any suggestions as to direction and magnitude?

> Are you seeing errors on the T1?

I'll look, when on site.

joe a.



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Re: [asterisk-users] Too much silence, perceived delay

[Stephen Bosch]

Joe Acquisto wrote:
> In a system connected to a verizon T1, Digium TE411P (quad T1 echo
> cancellation), client is complaining it is "too quiet".
> 
> The complaint regards calls over the T1, not in house SIP only calls.
> 
> Their description indicates they want some earpiece feedback of
> themselves speaking.  Also, they complain that it takes several
> seconds (3-4) for the other party to respond.  That is kind of
> subjective, I guess.

Have you been able to test this yourself? (Three to four seconds seems
inordinately long. That's as bad as a satellite link.)

Have you tried tinkering with the gain settings? Adjusting the gain can
impact sidetone, which might improve the call experience.

Are you seeing errors on the T1?

(A start, anyway)

-Stephen-





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Re: [asterisk-users] Date Wise Recordings

[Steve Edwards]

On Tue, 10 Apr 2007, Faisal Ashraf wrote:


I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's recordings
should go to next date folder.


Using the STRFTIME function to build your file name should get you close.

Reading the documentation should fill in the blanks :)

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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RE: [asterisk-users] Play audio and continue to next priority beforeaudio ends...

[Alejandro Mejía]

Thanks Steve.
I'll try what you suggest.

Cheers! 

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Steve Edwards
Enviado el: Lunes, 09 de Abril de 2007 04:08 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Play audio and continue to next priority
beforeaudio ends...

On Mon, 9 Apr 2007, Alejandro Mejma wrote:

> I would like to know how to playback an audio file to the caller, and 
> while it's played asterisk to continue executing the next priorities 
> on extensions.conf That's not the case when using "playback" command, 
> because the next priority is executed until the audio file ends 
> playing. I want to evaluate some variables while caller hears the 
> audio file.

I solved this in my application (adult chat billed to a credit card) by
writing an AGI that created a thread to play the audio ("Please wait while
your card is being verified") while the AGI mainline queried the credit card
processor. Most of the time I get the cc response before the audio finishes
so the authorization appears to be instantaneous.

The "key" was that the mainline cannot interact with Asterisk (i.e. 
"verbose" while the audio thread is executing.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] Too much silence, perceived delay

[Joe Acquisto]

In a system connected to a verizon T1, Digium TE411P (quad T1 echo 
cancellation), client is complaining it is "too quiet".

The complaint regards calls over the T1, not in house SIP only calls.

Their description indicates they want some earpiece feedback of themselves 
speaking.  Also, they complain that it takes several seconds (3-4) for the 
other party to respond.  That is kind of subjective, I guess.

Suggestions?

joe a.

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[asterisk-users] Date Wise Recordings

[Faisal Ashraf]

Hi,

I like my recordings to go to date wise folder i mean to say that for
example today is 20070409 so all recordings should go directly to that
folder instead of one folder for whole month. and then next day's recordings
should go to next date folder.

so how can i do that my current monitor context is like...

exten => _1NXXNXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)




Faisal
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Re: [asterisk-users] Play audio and continue to next priority before audio ends...

[Steve Edwards]

On Mon, 9 Apr 2007, Alejandro Mej?a wrote:


I would like to know how to playback an audio file to the caller, and while
it's played asterisk to continue executing the next priorities on
extensions.conf
That's not the case when using "playback" command, because the next priority
is executed until the audio file ends playing. I want to evaluate some
variables while caller hears the audio file.


I solved this in my application (adult chat billed to a credit card) by 
writing an AGI that created a thread to play the audio ("Please wait while 
your card is being verified") while the AGI mainline queried the credit 
card processor. Most of the time I get the cc response before the audio 
finishes so the authorization appears to be instantaneous.


The "key" was that the mainline cannot interact with Asterisk (i.e. 
"verbose" while the audio thread is executing.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
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[asterisk-users] Play audio and continue to next priority before audio ends...

[Alejandro Mejía]

Hello list members.
 
I would like to know how to playback an audio file to the caller, and while
it's played asterisk to continue executing the next priorities on
extensions.conf
That's not the case when using "playback" command, because the next priority
is executed until the audio file ends playing. I want to evaluate some
variables while caller hears the audio file.
 
Any ideas?
 
Thanks in advance,
 
Alejandro
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Re: [asterisk-users] Polycom 330/320

[Jessee J Holmes]

Mike,

I don't have much information, except they are due for shipment soon  
(mid to end of April to distribution from Polycom). We've demoed a  
couple and I personally believe they'll be a tough phone to find in  
stock for the first few months their released. Demand on these from  
what I'm seeing right now is very, very high. I think they are a  
great addition to the family and most importantly  they have FULL  
DUPLEX SPEAKERPHONE! :)


550's are released products though.

Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Apr 9, 2007, at 3:55 PM, Mike wrote:


Ah, thanks.  I didn't realize this.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Nickerson
Sent: Monday, April 09, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 330/320

Mike [EMAIL PROTECTED] wrote:


How do you guys like the 330 and 320?


Mike,

As far as I am aware, neither of these handsets are presently  
shipping from
Polycom, so most people's experience will be limited to PDF  
brochures and
pretty pictures. On the face of it, this looks like a good  
alternative to
the IP301 since it adds native 802.3af PoE support. Not sure yet  
exactly

where the pricing will slot in, however.

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)


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RE: [asterisk-users] TellMe Voice Recognition in Asterisk working..

[Dean Collins]

Hi Josh, fantastic implementation. 

It's a real shame tellme doesn't think that the 30,000+ Asterisk
installations don't warrant an ASP prepaid solution.

 

Regards,

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


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Josh Chaney
> Sent: Monday, 9 April 2007 4:38 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] TellMe Voice Recognition in Asterisk
working..
> 
> A couple of weekends ago I decided to see if I could get Asterisk to
> play nice with TellMe's VoiceXML studio. They provide the VoiceXML
> studio for free, and you can access it through SIP, so I thought this
> would be a fun and cheap way to integrate voice recognition into my
> IVR. I have posted a brief tutorial with code and examples on the
> voip-info.org wiki (
> http://www.voip-
> info.org/wiki/index.php?page=Add%20Voice%20Recognition%20to%20Asterisk
> ) as well as at my blog ( http://www.spinepunch.com ).
> 
> The code is a little rough, and I'm sure there is a better way of
> doing what I did, but this was easy and it worked for me. What's next
> on my to-do list is trying to cover up the TellMe jingle before it
> starts the VoiceXML app. If anyone would like to help clean up the
> code, or has a better way of interacting with the Asterisk manager,
> please let me know.
> 
>   Thanks,
>  -Josh
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RE: [asterisk-users] Polycom 330/320

[Mike]

Ah, thanks.  I didn't realize this.  

Mike 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Nickerson
Sent: Monday, April 09, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 330/320

Mike [EMAIL PROTECTED] wrote:

> How do you guys like the 330 and 320?

Mike,

As far as I am aware, neither of these handsets are presently shipping from
Polycom, so most people's experience will be limited to PDF brochures and
pretty pictures. On the face of it, this looks like a good alternative to
the IP301 since it adds native 802.3af PoE support. Not sure yet exactly
where the pricing will slot in, however.

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)


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[asterisk-users] TellMe Voice Recognition in Asterisk working..

[Josh Chaney]

A couple of weekends ago I decided to see if I could get Asterisk to
play nice with TellMe's VoiceXML studio. They provide the VoiceXML
studio for free, and you can access it through SIP, so I thought this
would be a fun and cheap way to integrate voice recognition into my
IVR. I have posted a brief tutorial with code and examples on the
voip-info.org wiki (
http://www.voip-info.org/wiki/index.php?page=Add%20Voice%20Recognition%20to%20Asterisk
) as well as at my blog ( http://www.spinepunch.com ).

The code is a little rough, and I'm sure there is a better way of
doing what I did, but this was easy and it worked for me. What's next
on my to-do list is trying to cover up the TellMe jingle before it
starts the VoiceXML app. If anyone would like to help clean up the
code, or has a better way of interacting with the Asterisk manager,
please let me know.

 Thanks,
-Josh
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Re: [asterisk-users] Polycom 330/320

[Darren Nickerson]

Mike [EMAIL PROTECTED] wrote:


How do you guys like the 330 and 320?


Mike,

As far as I am aware, neither of these handsets are presently shipping from 
Polycom, so most people's experience will be limited to PDF brochures and 
pretty pictures. On the face of it, this looks like a good alternative to 
the IP301 since it adds native 802.3af PoE support. Not sure yet exactly 
where the pricing will slot in, however.


-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)


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Re: [asterisk-users] T100P -->>> TE120P

[Carlos Chavez]

On Mon, 2007-04-09 at 12:50 -0700, Ritesh Agrawal wrote:
> Hi Folks,
> 
> I have a T100P ordered from Digium in the past. Its working perfectly
> fine but I now need to more the server to India (out-sourcing) and we
> will be getting an E1 line there.
> Does anyone know if T100P was capable (upgradeable) or swappable so
> that I could use it with the E1 lines instead? Any pointers to
> driver/firmware update would be 
> highly appreciated.
> 
No, that particular model is not able to use an E1.  You need a TE110P
which is able to select between both.  They used to have an E100P card
that was E1 only, both were replaced by the TE110P.

-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] ${QUEUESTATUS}

[Damon Estep]

There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when
they would be set;

 

If someone could correct errors with these definitions ot would be
appreciated;

 

TIMEOUT - the max time specified in the queue command elapsed, only
checked between retries so may not be 100% accurate.

FULL - the number of callers in the queues would exceed the maxlen=
value defined in queues.conf if another caller was added

JOINEMPTY - a call was sent to the queue but the queue had no members,
does not apply when using agentcallbacklogin since there could be
unavailable members defined but not available.

LEAVEEMPTY - the last agent was removed form the queue before alls calls
we handled, remong callers exit with this status, also acts differently
when there are only queue members that are unavaialbe

JOINUNAVAIL/LEAVEUNAVAIL - same as JOINEMPTY/LEAVEEMPTY, except that
there were still queue members, but all were status unavailable (logged
out)

 

So if a queue is made up of only callback agents (agentcallbacklogin)
then the queuestatus will never be joinempty or leaveempty

 

If the maxlen=0 then there eill never be a queuestatus of full

If there is no timeout in the queue command thee wll never be a
questatus of timeout

If there are no callback or static agents joinunavail/leaveunavail will
never apply.

 

 

Any corrections?

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[asterisk-users] Asterisk mini conference within IT360 in Toronto Apr30-May2nd

[Simon P. Ditner]

Hey all,

The Toronto AUG has been working with Clue.ca and IT360
(LinuxWorld/NetworkWorld), and has put together a mini-asterisk
conference within their larger conference:

  http://www.it360.ca/asterisk.cfm

If you're interested, as an 'association' we get 25% off the listed
prices. Our dicount code is: "A101", and our association name is:
"Asterisk User Group". Early bird rates end Wednesday April 11th.

Cheers,
spd
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[asterisk-users] T100P -->>> TE120P

[Ritesh Agrawal]

Hi Folks,

I have a T100P ordered from Digium in the past. Its working perfectly fine
but I now need to more the server to India (out-sourcing) and we will be
getting an E1 line there.
Does anyone know if T100P was capable (upgradeable) or swappable so that I
could use it with the E1 lines instead? Any pointers to driver/firmware
update would be
highly appreciated.

Thanks.
R
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Re: [asterisk-users] Privacy Manager w/ No Recording

[Peder @ NetworkOblivion]

I just opened 0009509 and used "Explicit Call Acceptance" as the name.

Ben Klang wrote:

On Monday 09 April 2007 02:48:32 pm Steve Murphy wrote:

On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote:

Is there a way to use privacy manager without requiring the user to
enter their name?  Essentially I am just looking for a way to force the
called user to enter "1" to accept the call.  I don't need a name
recording.  I want a call to come in, a message to be played, music on
hold, call out to the called party, then enter "1" to accept, "2" to
reject.

An interesting concept. File an enhancement request on bugs.digium.com,
and assign it to me, if you can; I'll look into it.
I wrote a patch for Asterisk 1.0 and 1.2 which implements this as an option to 
app_dial.  I called it Explicit Call Acceptance.  I believe others have done 
similar things both at the application level and at the dialplan level.  If 
you're interested, I'd be willing to resubmit my patch for consideration.


/BAK/


--

Network stuff you didn't know
http://www.networkoblivion.com

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Re: [asterisk-users] How well does a celldock work with Asterisk?

[Joe Acquisto]

Stephen Bosch <[EMAIL PROTECTED]> Wrote on: 4/9/2007 2:16 PM:
> Joe Acquisto wrote:
>>> Sometimes it's just a matter of finding a clean pair in the cable. Have
>>> you tried asking Verizon to fix the problem?
>> 
>> Don't get me started.  That's how I "know so much" about the situation.
>> They seem disinclined to address the matter, except with "happy talk" about
>> FIOS in my future.   "Soon".   Right after the metro areas are done.  Right.
>> 
>> The only fiber around here will be in my diet.
> 
> Hehe.
> 
> Are you in a rural area?
> 
> -Stephen-

Yes.  Woods, Hills and Dales, Bears, Deer and that sort of thing.  

It's an adventure.   But not that rural.   I am only about 500 ft beyond the 
max loop length, for DSL, and 1/4 mile from a Time Warner cable run.   But, are 
they flexible?   Will they work to make a customer happy?

Only in America.  (that used to mean something much different).

joe a.

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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 34

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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Re: [asterisk-users] Privacy Manager w/ No Recording

[Ben Klang]

On Monday 09 April 2007 02:48:32 pm Steve Murphy wrote:
> On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote:
> > Is there a way to use privacy manager without requiring the user to
> > enter their name?  Essentially I am just looking for a way to force the
> > called user to enter "1" to accept the call.  I don't need a name
> > recording.  I want a call to come in, a message to be played, music on
> > hold, call out to the called party, then enter "1" to accept, "2" to
> > reject.
> An interesting concept. File an enhancement request on bugs.digium.com,
> and assign it to me, if you can; I'll look into it.
I wrote a patch for Asterisk 1.0 and 1.2 which implements this as an option to 
app_dial.  I called it Explicit Call Acceptance.  I believe others have done 
similar things both at the application level and at the dialplan level.  If 
you're interested, I'd be willing to resubmit my patch for consideration.

/BAK/
-- 
Ben Klang
Alkaloid Networks
404.475.4850
[EMAIL PROTECTED]
http://projects.alkaloid.net
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Re: [asterisk-users] Re: incoming zaptel calls fail

[Kevin P. Fleming]

Robert La Ferla wrote:
> Neglected to mention the host operating system:
> 
> Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686 i686
> i386 GNU/Linux

You also neglected to mention the version of Asterisk you are running;
'latest SVN' means nothing when there are 20+ branches of Asterisk on
our SVN server.
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[asterisk-users] OT: But telephony related and funny

[Dean Collins]

Want to speak French? Just pick up the phone
 
A new mobile-phone service provides users with pictures of fictitious
lovers, and even fake conversations that let you pretend you're speaking
French. "So many people make fake phone calls to impress others, I
thought I'd make it easier, with better material," Mobile Faker founder
Cindy Lundin Mesaros said. New York Post
  (free
registration)

 

 

 

 

This was in my inbox from CTIA Smartbrief - lol, I thought it funny
enough to post.
>From what I can gather ith reads the lines to you to repeat so people
think you can speak french.

 

Cheers,

 

Dean

 

 

 

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Re: [asterisk-users] Privacy Manager w/ No Recording

[Steve Murphy]

On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote:
> Is there a way to use privacy manager without requiring the user to 
> enter their name?  Essentially I am just looking for a way to force the 
> called user to enter "1" to accept the call.  I don't need a name 
> recording.  I want a call to come in, a message to be played, music on 
> hold, call out to the called party, then enter "1" to accept, "2" to 
> reject.
> 
> Peder

Peder--

An interesting concept. File an enhancement request on bugs.digium.com,
and assign it to me, if you can; I'll look into it.

murf

-- 
Steve Murphy
Software Developer
Digium


smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] How well does a celldock work with Asterisk?

[Stephen Bosch]

Joe Acquisto wrote:
>> Sometimes it's just a matter of finding a clean pair in the cable. Have
>> you tried asking Verizon to fix the problem?
> 
> Don't get me started.  That's how I "know so much" about the situation.
> They seem disinclined to address the matter, except with "happy talk" about
> FIOS in my future.   "Soon".   Right after the metro areas are done.  Right.
> 
> The only fiber around here will be in my diet.

Hehe.

Are you in a rural area?

-Stephen-
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Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 33

[Alex Robar]

David,

It's not US format. He's away April 4th through April 11th. There was a big
discussion about FB and his absence on this list a few days ago.

Alex

On 4/9/07, David Boyd <[EMAIL PROTECTED]> wrote:


Could someone please remove this person from the list. It seems that the
person is saying they will be away for (9) nine months, with their
auto-reply set.

dave


On Mon, 2007-04-09 at 18:26 +0200, [EMAIL PROTECTED] wrote:
> Je suis absent du  2/04/2007 au 11/04/2007.
>
> Je répondrai à votre message dès mon retour. Pour toute urgence,
contacter
> Emmanuelle Parache Moga ou Cédric Buzay.
>
>
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--
Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 33

[David Boyd]

Could someone please remove this person from the list. It seems that the
person is saying they will be away for (9) nine months, with their
auto-reply set.

dave


On Mon, 2007-04-09 at 18:26 +0200, [EMAIL PROTECTED] wrote:
> Je suis absent du  2/04/2007 au 11/04/2007.
> 
> Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
> Emmanuelle Parache Moga ou Cédric Buzay.
> 
> 
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Re: [asterisk-users] "remote" SIP, no audio, or one way audio.

[Joe Acquisto]

Hi.

Is there a way to isolate what shows on CLI to just the conversation with that 
extension?   There appears to be a lot of stuff unrelated to this extension.

Packet traces are not out of the question, but cannot be done today.

joe a.

"Yossi Ben Hagai" <[EMAIL PROTECTED]> Wrote: 4/9/2007 12:56 PM:
> Hi Joe,
> 
> The debug trace you've enclosed is a NOTIFY message sent from * for the
> message waiting feature - and is not related to the call.
> You can however tell that something is wrong since the message is being
> retransmitted since the server didn't receive 200 OK in reply - while it
> could be due to the client being offline or not supporting this feature 
> It
> could imply a NAT issue so try to recheck your NAT configs.
> 
> can you post a full trace (starting with the INVITE message)? also you 
> can
> try to run a sniffer trace on the client side to see if it 
> receives/sends
> the messages correctly.
> 
> Joss.
> 
> On 4/9/07, Joe Acquisto <[EMAIL PROTECTED]> wrote:
>>
>> I never get this far, apparently.   While the connection seems to be made,
>> and calls can be "completed" (rings, answers) there is no audio.   On CLI, I
>> can see what appears to be call being made and connected.  These are x-lite
>> phones (for testing, one hopes) there appears to be no codec selection
>> available.
>>
>> I see no CODEC dialog.  What I see is six iterations of the below:
>>
>> . . . .
>> ---
>>
>> Retransmitting #6 (NAT) to xx.xx.xx.xx:64909:
>> NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
>> Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport
>> From: "n"> To: "n";tag=9c58a77e
>> Contact: 
>> Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE.
>> CSeq: 102 NOTIFY
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Event: message-summary
>> Content-Type: application/simple-message-summary
>> Subscription-State: terminated;reason=timeout
>> Content-Length: 0
>> -
>>
>> Does this imply anyting to anyone?
>>
>> Call can be made, after this.
>>
>> joe a.
>>
>> **
>> dave cantera <[EMAIL PROTECTED]> Wrote: 4/7/2007 3:53 PM:
>> > joe,
>> > when I have problems with audio and other connections seem to work, I
>> > always look for a codec incompatibility...  use  'sip set debug peer
>> > '  and look for the codec handshaking... make sure both
>> > extensions have a compatible codec choice...
>> > daveC
>> >
>> > Using INVITE request as basis request - [EMAIL PROTECTED] 
>> > Found user '401'
>> > Found RTP audio format 0
>> > Found RTP audio format 8
>> > Found RTP audio format 3
>> > Found RTP video format 99
>> > Peer audio RTP is at port 192.168.15.100:5004
>> >
>> > *Found description format PCMU for ID 0
>> > Found description format PCMA for ID 8
>> > Found description format GSM for ID 3
>> > Found description format H264 for ID 99
>> >
>> > *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer -
>> > audio=0x2e
>> > (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e
>> > (gsm|ulaw|alaw|h264)
>> >
>> > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
>> > (nothing), combined - 0x0 (nothing)
>> > Peer audio RTP is at port 192.168.15.100:5004
>> > Peer video RTP is at port 192.168.15.100:5006
>> > Looking for 404 in inbound-video (domain sip3701.ibsonecall.com)
>> > list_route: hop: 
>> >
>> >
>> >
>> > Joe Acquisto wrote:
>> >> Steve Totaro <[EMAIL PROTECTED]> Wrote: 4/4/2007 8:44 PM:
>> >>
>> >>> Joe Acquisto wrote:
>> >>>
>>  Attempts to do SIP thru firewall (IPCop) are unsuccessful.  using
>> x-lite
>>  softphones, for eval/testing.  They do get registered, and can call
>> each
>>  other, but mostly get no audio, sometimes one way audio.
>> 
>>  Suggestions/fixes?
>> 
>>  joe a.
>> 
>> 
>> >>> Is there NAT on both sides?  Are you using qualify?  Paint a clearer
>> >>> picture.
>> >>>
>> >>>
>> >>
>> >>
>> >> Sorry, I missed your reply, till now.
>> >>
>> >> --switch
>> >>  |  | |phones
>> >>  |  |-asterisk box
>> >>
>> >>
>> |---IPcop|---internet-|-home/remote-office--
>> >> --|sip phone
>> >>
>> >> |-ditto
>> >>
>> >> Hope that is intelligible.
>> >>
>> >> joe a
>> >>
>> >> ___
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>> >> asterisk-users mailing list
>> >> To UNSUBSCRIBE or update options visit:
>> >>http://lists.digium.com/mailman/listinfo/asterisk-users 
>> >>
>> >>
>> >>
>> >>
>> >
>> > --
>> > Building Strong Relationships w/ Intelligent Customer Service
>> > --
>> >
>> > Interlocking Business Solutions, LLC
>> > 856-380-0894 x5000
>> >
>> >
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>>
>>
>> _

[asterisk-users] Re: DTMF auto detection bug?

[Arik Raffael Funke]

Joshua Colp wrote:

Arik Raffael Funke wrote:
The auto setting also does not encompass the info 
DTMF option for sending.


Thanks. I was not aware of this.

Ragards,
- Arik

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Re: [asterisk-users] Received mini frame before first full voice frame

[Tim Panton]

On 9 Apr 2007, at 17:10, voiplist wrote:


Can someone give me a little detail as to what this error message
means and why it may be occuring?

I keep seeing tons of these roll by on the CLI on one of our systems.

Thanks!

Apr  9 11:05:40 WARNING[19263]: chan_iax2.c:7538 socket_read:  
Received mini fra

me before first full voice frame


It means that the audio (mini-frames) for a call has started arriving  
before the
call was fully set up (by a Voice frame). Your asterisk isn't sure  
what format the

media is in, so is dumping it for now.

What causes this usually is a lost Voice frame - the far end will  
retry after a while

and if that arrives all will be well from then on.



Tim Panton

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] "remote" SIP, no audio, or one way audio.

[Yossi Ben Hagai]

Hi Joe,

The debug trace you've enclosed is a NOTIFY message sent from * for the
message waiting feature - and is not related to the call.
You can however tell that something is wrong since the message is being
retransmitted since the server didn't receive 200 OK in reply - while it
could be due to the client being offline or not supporting this feature It
could imply a NAT issue so try to recheck your NAT configs.

can you post a full trace (starting with the INVITE message)? also you can
try to run a sniffer trace on the client side to see if it receives/sends
the messages correctly.

Joss.

On 4/9/07, Joe Acquisto <[EMAIL PROTECTED]> wrote:


I never get this far, apparently.   While the connection seems to be made,
and calls can be "completed" (rings, answers) there is no audio.   On CLI, I
can see what appears to be call being made and connected.  These are x-lite
phones (for testing, one hopes) there appears to be no codec selection
available.

I see no CODEC dialog.  What I see is six iterations of the below:

. . . .
---

Retransmitting #6 (NAT) to xx.xx.xx.xx:64909:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport
From: "n";tag=9c58a77e
Contact: 
Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: terminated;reason=timeout
Content-Length: 0
-

Does this imply anyting to anyone?

Call can be made, after this.

joe a.

**
dave cantera <[EMAIL PROTECTED]> Wrote: 4/7/2007 3:53 PM:
> joe,
> when I have problems with audio and other connections seem to work, I
> always look for a codec incompatibility...  use  'sip set debug peer
> '  and look for the codec handshaking... make sure both
> extensions have a compatible codec choice...
> daveC
>
> Using INVITE request as basis request - [EMAIL PROTECTED]
> Found user '401'
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 3
> Found RTP video format 99
> Peer audio RTP is at port 192.168.15.100:5004
>
> *Found description format PCMU for ID 0
> Found description format PCMA for ID 8
> Found description format GSM for ID 3
> Found description format H264 for ID 99
>
> *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer -
> audio=0x2e
> (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e
> (gsm|ulaw|alaw|h264)
>
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
> (nothing), combined - 0x0 (nothing)
> Peer audio RTP is at port 192.168.15.100:5004
> Peer video RTP is at port 192.168.15.100:5006
> Looking for 404 in inbound-video (domain sip3701.ibsonecall.com)
> list_route: hop: 
>
>
>
> Joe Acquisto wrote:
>> Steve Totaro <[EMAIL PROTECTED]> Wrote: 4/4/2007 8:44 PM:
>>
>>> Joe Acquisto wrote:
>>>
 Attempts to do SIP thru firewall (IPCop) are unsuccessful.  using
x-lite
 softphones, for eval/testing.  They do get registered, and can call
each
 other, but mostly get no audio, sometimes one way audio.

 Suggestions/fixes?

 joe a.


>>> Is there NAT on both sides?  Are you using qualify?  Paint a clearer
>>> picture.
>>>
>>>
>>
>>
>> Sorry, I missed your reply, till now.
>>
>> --switch
>>  |  | |phones
>>  |  |-asterisk box
>>
>>
|---IPcop|---internet-|-home/remote-office--
>> --|sip phone
>>
>> |-ditto
>>
>> Hope that is intelligible.
>>
>> joe a
>>
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>>
>>
>>
>
> --
> Building Strong Relationships w/ Intelligent Customer Service
> --
>
> Interlocking Business Solutions, LLC
> 856-380-0894 x5000
>
>
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Re: [asterisk-users] How well does a celldock work with Asterisk?

[Joe Acquisto]

. . . 
>> Can't be worse than my POTS lines.   The cable runs here are about 30
>> years old, and run underground, supposedly, where crossing a
>> government right of way.   This run is ancient, as well.
>> Supposedly, during wet weather, this becomes a grounding problem.
>> Certainly the audio quality deteriorates to the point of being almost
>> unusable, during inclement weather and clears up when dry.
> 
> The cable sleeve is probably breached and I'll bet the waterproofing
> jelly has leaked out. If, along with that, you have any cracked
> insulation on the individual conductors, the results are predictable.
> This is a common problem on older runs.
> 
> Sometimes it's just a matter of finding a clean pair in the cable. Have
> you tried asking Verizon to fix the problem?

Don't get me started.  That's how I "know so much" about the situation.
They seem disinclined to address the matter, except with "happy talk" about
FIOS in my future.   "Soon".   Right after the metro areas are done.  Right.

The only fiber around here will be in my diet.

joe a.

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Re: [asterisk-users] sip_header=value?

[Karl J. Vesterling]

Apparently it sets a SIP_HEADER variable named "Call-Info" to a value of
"answer-after=0" effectively telling the Sipura to answer the call and
put it through to speakerphone.

I will say that extensions.ael is a bit different from regular line
based extensions.conf in that I seem to have to escape all sorts of
stuff with the \ character that I don't have to in extensions.conf

Back to work, I'll check in on this thread later this evening.


Rizwan Hisham wrote:
> I dont understand it
>
> Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
>
> whats it doing here?
>
>
> On 4/9/07, *Karl J. Vesterling* < [EMAIL PROTECTED]
> > wrote:
>
> I struggled with this one too, try this:
> Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
>
> I use the above for intercom w/ Sipura SPA-941 and it works.
> Asterisk 1.2.17 / extensions.ael
>
>
>
> Rizwan Hisham wrote:
>> I have tried it, it doesnt work
>>
>> On 4/9/07, *Hermann Wecke* <[EMAIL PROTECTED]
>> > wrote:
>>
>> Rizwan Hisham wrote:
>> > is there anyway i can set SIP_HEADER(To) to the value i like?
>>
>> If voip-info is correct, you can read, but you can't change.
>> http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
>> 
>> 
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>>
>>
>>
>> -- 
>> Regards
>> Rizwan Hisham
>> Software Engineer
>> 
>>
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>>   
>
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>
>
>
>
> -- 
> Regards
> Rizwan Hisham
> Software Engineer
> 
>
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[asterisk-users] trouble recording calls

[ahester]

Hi all,

I am having the following trouble with recording calls:
When calls come into the support line did number, the call starts to
record on the first queue, but appears to hang up when the call actually
connects to the engineer (ie I see "got hangup request" on the cli and
then mixmonitor ends.)  I am guessing this has to do with the announce
file that is played to the engineer before the call is connected.  It
seems that if the call rolls to the next queue because of timeout,
asterisk doesn't even try to record it. (I don't see any mixmonitor on
the cli for the next queue). 

I would appreciate any help with this.  I have to have all calls
recorded and I have to do announcements so that the callee knows how to
answer the phone.

Thanks,
Andy


The configs are as below:

>From extensions.conf:

#after various menu stuff, send to support
exten => 214xxx,13,SetGlobalVar(ORIGIN=support)
exten => 214xxx,14,Queue(support1|tr|||10)
exten => 214xxx,15,Queue(support2|tr|||)

#dial command for sip extensions that are in the queues
exten =>
_72XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten => _72XXX,2,Dial(SIP/${EXTEN})
exten =>
_73XXX,1,MixMonitor(${ORIGIN}/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
exten => _73XXX,2,Dial(SIP/${EXTEN})


queues from queues.conf:

[support1]
; Support call queue
announce = 16
strategy = rrmemory
timeout = 15
retry = none
wrapuptime=15
announce-frequency = 0
joinempty = no
leavewhenempty = yes
member => Agent/2008
member => Agent/2009
member => Agent/2014
member => Agent/2015
member => Agent/2017
member => Agent/2018
member => Agent/2019
member => Agent/3520
member => Agent/3521
member => Agent/3522
member => Agent/3524
member => Agent/3529

[support2]
; Support2 call queue
announce = 16
strategy = ringall
announce-frequency = 0
; Added below for testing because the second queue was not even trying
to record
; according to the asterisk console (still doesn't)
Set(MONITOR_FILENAME=support/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav)
monitor-format = wav
monitor-join = yes
joinempty = yes
member => SIP/72008
member => SIP/72009


 

-- 
Andy Hester
Network Engineer
Architel

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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 33

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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Re: [asterisk-users] Asterisk installation issue - CLI showing 0 active channels

[Vijay Gaur]

Yes when I plug my phone to vonage adapter it rings fine.
I will run and send you the output soon.
Thanks

Vijay
On 4/9/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:


Vijay Gaur wrote:
> I hear ring few times and then it goes to voice mail. Looks like call is
> not going to asterisk. My regular phone attached to that line works
fine.

When you plug the phone into the port on the Vonage ATA that you're
using to connect to Asterisk, the phone rings when you call the number?

Here's my next question: What does cat /proc/interrupts show on the
Asterisk server? Run that and post the output.

-Stephen-
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[asterisk-users] Polycom 330/320

[Mike]

How do you guys like the 330 and 320?  I've been looking at this as my
"standard" phone, since it's relatively cheaper than the 501 which is the
phone I currently push with my PBX systems.  Most of my customers do not use
more than one line per phone, so having 3 lines on the 501 is not
necessarily useful. 
 
Also, I read that the phone offers TLS security.  What does that mean?  I
understand Asterisk does not, but is this something that could be possible
with futur asterisk developement?
 
Mike
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[asterisk-users] Received mini frame before first full voice frame

[voiplist]

Can someone give me a little detail as to what this error message
means and why it may be occuring?

I keep seeing tons of these roll by on the CLI on one of our systems.

Thanks!

Apr  9 11:05:40 WARNING[19263]: chan_iax2.c:7538 socket_read: Received mini fra
me before first full voice frame
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Re: [asterisk-users] Verizon-Vonage Lawsuit

[Christopher Chan]

Just curious,

Christopher, what is a chicken boner?




Sorry, that's anti-spammer jargon for spammer. I used to be a mail admin 
for an outfit that handles over 40 million mailboxes and over 200 
million email transactions daily. Guess what composed the majority of 
the daily 200 million transactions...and their origin...and the 
'software' used


http://www.spamfaq.net/terminology.shtml#chickenboner

http://www.netlingo.com/lookup.cfm?term=chicken-boner
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Re: [asterisk-users] Asterisk installation issue - CLI showing 0 active channels

[Stephen Bosch]

Vijay Gaur wrote:
> I hear ring few times and then it goes to voice mail. Looks like call is
> not going to asterisk. My regular phone attached to that line works fine.

When you plug the phone into the port on the Vonage ATA that you're
using to connect to Asterisk, the phone rings when you call the number?

Here's my next question: What does cat /proc/interrupts show on the
Asterisk server? Run that and post the output.

-Stephen-
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Re: [asterisk-users] How well does a celldock work with Asterisk?

[Stephen Bosch]

Joe Acquisto wrote:
> . . .
>> I have a Dock-n-Talk at home I use to connect my motorola V60i via
>> a cable so I can't comment on bluetooth. I needed it because for
>> some reason I can only get good cell reception in my bedroom. It
>> works well enough. You can certainly tell you are talking over a
>> cell connection and not a POTS line (it's a little noisier) and now
>> and then I have slight echo. Caller ID is passed through (number
>> only) as well. All in all I'm happy with the purchase.
>> 
>> -Dave
> 
> Can't be worse than my POTS lines.   The cable runs here are about 30
> years old, and run underground, supposedly, where crossing a
> government right of way.   This run is ancient, as well.
> Supposedly, during wet weather, this becomes a grounding problem.
> Certainly the audio quality deteriorates to the point of being almost
> unusable, during inclement weather and clears up when dry.

The cable sleeve is probably breached and I'll bet the waterproofing
jelly has leaked out. If, along with that, you have any cracked
insulation on the individual conductors, the results are predictable.
This is a common problem on older runs.

Sometimes it's just a matter of finding a clean pair in the cable. Have
you tried asking Verizon to fix the problem?

In these parts, Telus (incidentally, 33% owned by Verizon) will try
switching pairs in a segment if you cajole them, and that will sometimes
solve it.

(Either that or there's a water-logged pedestal somewhere)

-Stephen-
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Re: [asterisk-users] Asterisk installation issue - CLI showing 0 active channels

[Vijay Gaur]

I hear ring few times and then it goes to voice mail. Looks like call is not
going to asterisk. My regular phone attached to that line works fine.
Regards,
Vijay


On 4/9/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:


Vijay Gaur wrote:
> Hi Stephen,
>   I made the call from outside phone(cellphone) to my vonage number.
> Call went to the voicemail. My extension.conf has
> [incoming]
> exten=>s,1,Answer() and more lines.
>
> I assume core show channels should show 1 active channel. Its showing 0.

Your calls are not even going into Asterisk.

When it goes to voice mail, do you hear it ring a few times first, or
does it go directly to voicemail?

Does a regular telephone plugged into the Vonage adapter work?

-Stephen-
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Re: [asterisk-users] sip_header=value?

[Rizwan Hisham]

Guys, i have solved my problem thru different means, i only need to pass the
dnid when the user is using asterisk to regiter as a peer. so...here is
my solution

exten=> 123,1,Gotoif($["${SIPPEER(abc:useragent)}" = "Asterisk PBX"]?20:30)

exten=> 123,20,Dial(SIP/[EMAIL PROTECTED],,Tt)
exten=> 123,21,hangup

exten=> 123,30,Dial(SIP/abc,,Tt)
exten=> 123,31,hangup

it works as i want it to work now

On 4/9/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote:


I dont understand it

Set(__SIPADDHEADER=Call-Info:\;answer-after=0);

whats it doing here?


On 4/9/07, Karl J. Vesterling < [EMAIL PROTECTED]> wrote:
>
>  I struggled with this one too, try this:
> Set(__SIPADDHEADER=Call-Info:\;answer-after=0);
>
> I use the above for intercom w/ Sipura SPA-941 and it works.
> Asterisk 1.2.17 / extensions.ael
>
>
>
> Rizwan Hisham wrote:
>
> I have tried it, it doesnt work
>
> On 4/9/07, Hermann Wecke <[EMAIL PROTECTED]> wrote:
> >
> > Rizwan Hisham wrote:
> > > is there anyway i can set SIP_HEADER(To) to the value i like?
> >
> > If voip-info is correct, you can read, but you can't change.
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> Regards
> Rizwan Hisham
> Software Engineer
>
> --
>
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>


--
Regards
Rizwan Hisham
Software Engineer





--
Regards
Rizwan Hisham
Software Engineer
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Re: [asterisk-users] Asterisk installation issue - CLI showing 0 active channels

[Stephen Bosch]

Vijay Gaur wrote:
> Hi Stephen,
>   I made the call from outside phone(cellphone) to my vonage number.
> Call went to the voicemail. My extension.conf has
> [incoming]
> exten=>s,1,Answer() and more lines.
>  
> I assume core show channels should show 1 active channel. Its showing 0.

Your calls are not even going into Asterisk.

When it goes to voice mail, do you hear it ring a few times first, or
does it go directly to voicemail?

Does a regular telephone plugged into the Vonage adapter work?

-Stephen-
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RE: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

[Bobby Crawford]

The higher price on the Sangoma is for hardware echo cancellation.  There
should be a model (A20400) that doesn't have the echo cancellation and it
probably is less expensive than the Digium card.

Bobby

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze
Sent: Monday, April 09, 2007 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

On Apr 9, 2007, at 9:29 AM, William Moore wrote:

> On 4/9/07, Jim Freeze <[EMAIL PROTECTED]> wrote:
>> > Or a new Digium TDM880B replacing the old TDM40B for only one  
>> IRQ...
>> Do you know if this board will fit in a 2U machine?
>
> The TDM800P is about the same height as the TDM400P and is about an
> inch longer, so you should have no problem putting it in the same slot
> as the TDM400 was in.

I really appreciate all the info. I think I am going to go the route  
of an 8 FXS card,
but still need to decide between Digium and Sangoma.

Are there any recommendations between:

  TDM880B - $679
  A20400D  - $906

Also, why the higher price on the Sangoma?


Jim

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Re: [asterisk-users] Asterisk installation issue - CLI showing 0 active channels

[Vijay Gaur]

Hi Stephen,
 I made the call from outside phone(cellphone) to my vonage number. Call
went to the voicemail. My extension.conf has
[incoming]
exten=>s,1,Answer() and more lines.

I assume core show channels should show 1 active channel. Its showing 0.

Thanks a lot again Stephen.
Regards,
Vijay




On 4/9/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:


Vijay Gaur wrote:
>
>
> Hi All,
>I would appreciate a lot if you could help me. I have installed
> Asterisk 1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have
> also installed 1 FXO port card: Digium TDM400P.
> After loading zaptel driver I could see my digium card's led glow green.
> Tested with zttool that its in OK state. I have configured fxsks=4 in
> zaptel.conf(channel 4 cause FXO module is on port 4). I have also
> configured channel 4 in incoming context in zapata.conf.
>
> After all configuration I started asterisk with vgc and made a call,
> but I didn't see any call log on asterisk console.

We need much more information.

>From *where* did you make a call? Did you dial from another location
into the Asterisk number (e.g., did you dial your Vonage number from
outside?), or did you call out from your Asterisk server using an
internal extension?

Does the call succeed? Do you hear ringing, or a fast busy, or does the
call go to voice mail?

What is in your dialplan (extensions.conf)?

> When I did core show channels its showing 0 active channels. Looks like
> I did something wrong in my config.
>
> Also I dont see red-alarm in zttool if I take out pstn line.

I don't think the TDM400P cards do media detection :) You're probably
looking for the problem in the wrong place -- but answer the above
questions first and we'll see what we can figure out.

-Stephen-


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Re: [asterisk-users] sip_header=value?

[Rizwan Hisham]

I dont understand it

Set(__SIPADDHEADER=Call-Info:\;answer-after=0);

whats it doing here?


On 4/9/07, Karl J. Vesterling <[EMAIL PROTECTED]> wrote:


 I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);

I use the above for intercom w/ Sipura SPA-941 and it works.
Asterisk 1.2.17 / extensions.ael



Rizwan Hisham wrote:

I have tried it, it doesnt work

On 4/9/07, Hermann Wecke <[EMAIL PROTECTED]> wrote:
>
> Rizwan Hisham wrote:
> > is there anyway i can set SIP_HEADER(To) to the value i like?
>
> If voip-info is correct, you can read, but you can't change.
> http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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>



--
Regards
Rizwan Hisham
Software Engineer

--

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--
Regards
Rizwan Hisham
Software Engineer
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Re: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

[Jim Freeze]

On Apr 9, 2007, at 9:29 AM, William Moore wrote:


On 4/9/07, Jim Freeze <[EMAIL PROTECTED]> wrote:
> Or a new Digium TDM880B replacing the old TDM40B for only one  
IRQ...

Do you know if this board will fit in a 2U machine?


The TDM800P is about the same height as the TDM400P and is about an
inch longer, so you should have no problem putting it in the same slot
as the TDM400 was in.


I really appreciate all the info. I think I am going to go the route  
of an 8 FXS card,

but still need to decide between Digium and Sangoma.

Are there any recommendations between:

 TDM880B - $679
 A20400D  - $906

Also, why the higher price on the Sangoma?


Jim

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[asterisk-users] Privacy Manager w/ No Recording

[Peder @ NetworkOblivion]

Is there a way to use privacy manager without requiring the user to 
enter their name?  Essentially I am just looking for a way to force the 
called user to enter "1" to accept the call.  I don't need a name 
recording.  I want a call to come in, a message to be played, music on 
hold, call out to the called party, then enter "1" to accept, "2" to 
reject.


Peder

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Re: [asterisk-users] DTMF auto detection bug?

[Joshua Colp]

Arik Raffael Funke wrote:

Hi,

it seems that there is a bug in asterisk's dtmf mode autodetection. 
Assume following sip.conf:


[sipprovider]
disallow=all
allow=g726
dtmfmode=auto

DTMF does not work. It seems rfc2833 mode is chosen despite it being 
obvious that this cannot work!


Why could it not work? While RFC2833 is carried over the RTP stream it 
is not sent in the audio stream so it should work as long as the remote 
end is using it.



The following configuration is necessary to get DTMF to work: dtmfmode=info


It's always best to specify the dtmfmode in use as it controls both 
receiving and sending. The auto setting also does not encompass the info 
DTMF option for sending.


In my opinion, this behaviour is counter-intuitive. I am using asterisk 
1.2. In v. 1.4 does dtmfmode=auto still have the behaviour?


I do not believe it has been changed at all. If you have a method of 
better determining things please feel free to share it.



Cheers,
Arik



Joshua Colp
Software Developer
Digium, Inc.
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Re: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

[William Moore]

On 4/9/07, Jim Freeze <[EMAIL PROTECTED]> wrote:

> Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...
Do you know if this board will fit in a 2U machine?


The TDM800P is about the same height as the TDM400P and is about an
inch longer, so you should have no problem putting it in the same slot
as the TDM400 was in.
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Re: RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

[Jim Freeze]

On Apr 9, 2007, at 8:32 AM, <[EMAIL PROTECTED]> [EMAIL PROTECTED]> wrote:



Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...


Do you know if this board will fit in a 2U machine?

Thanks

Jim



Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jim  
Freeze

Envoyé : lundi 9 avril 2007 15:15
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Upgrade 4 to 8 Analog Lines Question


Hello

I have an office with a T1 that provides 4 (out of 8) analog PSTN  
lines thru

an adtran board. I want to add 4 more analog lines. Currently I have a
Digium TDM40B. I'm wondering what the best upgrade path is, where I  
define
'best' as the solution that is most likely to work without problems  
(like

interupt conflicts) and work with my current echo tuning .

I see my purchase options as follows:

1) TDM40B - use with the current TDM40B
2) Sangoma Remora A20200 - use with the current TDM40B
3) Sangoma Remora A20400 - replace the current TDM40B


Any info will be greatly appreciated.

Thanks

Jim


--
Jim Freeze
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Re: [asterisk-users] Verizon-Vonage Lawsuit

[Rob Hillis]

Only when the chicken is provided with sufficient stimulation.

Salvatore Giudice wrote:

I think it's a small, feather covered appendage.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Monday, April 09, 2007 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit

Just curious,

Christopher, what is a chicken boner?

  

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[asterisk-users] DTMF auto detection bug?

[Arik Raffael Funke]

Hi,

it seems that there is a bug in asterisk's dtmf mode autodetection. 
Assume following sip.conf:


[sipprovider]
disallow=all
allow=g726
dtmfmode=auto

DTMF does not work. It seems rfc2833 mode is chosen despite it being 
obvious that this cannot work!


The following configuration is necessary to get DTMF to work: dtmfmode=info

In my opinion, this behaviour is counter-intuitive. I am using asterisk 
1.2. In v. 1.4 does dtmfmode=auto still have the behaviour?


Cheers,
Arik

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[asterisk-users] Re: incoming zaptel calls fail

[Robert La Ferla]

Neglected to mention the host operating system:

Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686  
i686 i386 GNU/Linux


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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 32

[fb]

Je suis absent du  2/04/2007 au 11/04/2007.

Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.


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Re: [asterisk-users] sip_header=value?

[Karl J. Vesterling]

I struggled with this one too, try this:
Set(__SIPADDHEADER=Call-Info:\;answer-after=0);

I use the above for intercom w/ Sipura SPA-941 and it works.
Asterisk 1.2.17 / extensions.ael



Rizwan Hisham wrote:
> I have tried it, it doesnt work
>
> On 4/9/07, *Hermann Wecke* <[EMAIL PROTECTED]
> > wrote:
>
> Rizwan Hisham wrote:
> > is there anyway i can set SIP_HEADER(To) to the value i like?
>
> If voip-info is correct, you can read, but you can't change.
> http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
> ___
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>  --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Regards
> Rizwan Hisham
> Software Engineer
> 
>
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Re: [asterisk-users] Asterisk installation issue - CLI showing 0 active channels

[Stephen Bosch]

Vijay Gaur wrote:
> 
> 
> Hi All,
>I would appreciate a lot if you could help me. I have installed
> Asterisk 1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have
> also installed 1 FXO port card: Digium TDM400P.
> After loading zaptel driver I could see my digium card's led glow green.
> Tested with zttool that its in OK state. I have configured fxsks=4 in
> zaptel.conf(channel 4 cause FXO module is on port 4). I have also
> configured channel 4 in incoming context in zapata.conf.
>  
> After all configuration I started asterisk with vgc and made a call,
> but I didn't see any call log on asterisk console.

We need much more information.

>From *where* did you make a call? Did you dial from another location
into the Asterisk number (e.g., did you dial your Vonage number from
outside?), or did you call out from your Asterisk server using an
internal extension?

Does the call succeed? Do you hear ringing, or a fast busy, or does the
call go to voice mail?

What is in your dialplan (extensions.conf)?

> When I did core show channels its showing 0 active channels. Looks like
> I did something wrong in my config.
>  
> Also I dont see red-alarm in zttool if I take out pstn line.

I don't think the TDM400P cards do media detection :) You're probably
looking for the problem in the wrong place -- but answer the above
questions first and we'll see what we can figure out.

-Stephen-


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[asterisk-users] incoming zaptel calls fail

[Robert La Ferla]

Using the latest SVN of zaptel and asterisk, I can no longer receive  
incoming analog calls.  The caller just hears it ringing but Asterisk  
doesn't pick up.


I am seeing these error messages:

[Apr  9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID  
returned with error on channel 'Zap/2-1'
[Apr  9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID  
returned with error on channel 'Zap/2-1'
[Apr  9 09:39:34] WARNING[16580]: chan_zap.c:6470 ss_thread: CallerID  
returned with error on channel 'Zap/2-1'


and also:

 % dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: SVN-branch-1.4-r2397M
Zaptel Echo Canceller: MG2
Freshmaker version: 73
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Not installed
Module 3: Not installed
Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules)
Registered tone zone 0 (United States / North America)
Zaptel Transcoder support loaded
zaptel.c:764 (pid 3176: asterisk) got signal 8000
zaptel.c:764 (pid 16469: asterisk) got signal 8100

What could be the problem?  How can I fix it?

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Re: [asterisk-users] How well does a celldock work with Asterisk?

[Joe Acquisto]

. . . 
> 
> I have a Dock-n-Talk at home I use to connect my motorola V60i via a 
> cable so I can't comment on bluetooth. I needed it because for some 
> reason I can only get good cell reception in my bedroom. It works well 
> enough. You can certainly tell you are talking over a cell connection 
> and not a POTS line (it's a little noisier) and now and then I have 
> slight echo. Caller ID is passed through (number only) as well.
> All in all I'm happy with the purchase.
> 
> -Dave

Can't be worse than my POTS lines.   The cable runs here are about 30 years 
old, and run underground, supposedly, where crossing a government right of way. 
  This run is ancient, as well.   Supposedly, during wet weather, this becomes 
a grounding problem.  Certainly the audio quality deteriorates to the point of 
being almost unusable, during inclement weather and clears up when dry.

As I get fair reception, using Nextel, in this area . . .

joe a.

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Re: [asterisk-users] "remote" SIP, no audio, or one way audio.

[Joe Acquisto]

I never get this far, apparently.   While the connection seems to be made, and 
calls can be "completed" (rings, answers) there is no audio.   On CLI, I can 
see what appears to be call being made and connected.  These are x-lite phones 
(for testing, one hopes) there appears to be no codec selection available.

I see no CODEC dialog.  What I see is six iterations of the below:

. . . .
---

Retransmitting #6 (NAT) to xx.xx.xx.xx:64909:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport
From: "n";tag=9c58a77e
Contact: 
Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: terminated;reason=timeout
Content-Length: 0
-

Does this imply anyting to anyone?

Call can be made, after this.

joe a.

**
dave cantera <[EMAIL PROTECTED]> Wrote: 4/7/2007 3:53 PM:
> joe,
> when I have problems with audio and other connections seem to work, I 
> always look for a codec incompatibility...  use  'sip set debug peer 
> '  and look for the codec handshaking... make sure both 
> extensions have a compatible codec choice...
> daveC
> 
> Using INVITE request as basis request - [EMAIL PROTECTED] 
> Found user '401'
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 3
> Found RTP video format 99
> Peer audio RTP is at port 192.168.15.100:5004
> 
> *Found description format PCMU for ID 0
> Found description format PCMA for ID 8
> Found description format GSM for ID 3
> Found description format H264 for ID 99
> 
> *Capabilities: us - 0x2e (gsm|ulaw|alaw|h264), peer - 
> audio=0x2e 
> (gsm|ulaw|alaw|h264)/video=0x20 (h264), combined - 0x2e 
> (gsm|ulaw|alaw|h264)
> 
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 
> (nothing), combined - 0x0 (nothing)
> Peer audio RTP is at port 192.168.15.100:5004
> Peer video RTP is at port 192.168.15.100:5006
> Looking for 404 in inbound-video (domain sip3701.ibsonecall.com)
> list_route: hop: 
> 
> 
> 
> Joe Acquisto wrote:
>> Steve Totaro <[EMAIL PROTECTED]> Wrote: 4/4/2007 8:44 PM:
>>   
>>> Joe Acquisto wrote:
>>> 
 Attempts to do SIP thru firewall (IPCop) are unsuccessful.  using x-lite 
 softphones, for eval/testing.  They do get registered, and can call each 
 other, but mostly get no audio, sometimes one way audio.

 Suggestions/fixes?

 joe a.
   
   
>>> Is there NAT on both sides?  Are you using qualify?  Paint a clearer 
>>> picture.
>>>
>>> 
>>
>>
>> Sorry, I missed your reply, till now.
>>
>> --switch
>>  |  | |phones
>>  |  |-asterisk box
>>  
>> |---IPcop|---internet-|-home/remote-office--
>> --|sip phone
>> 
>> |-ditto
>>
>> Hope that is intelligible.
>>
>> joe a
>>
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>>
>>
>>   
> 
> -- 
> Building Strong Relationships w/ Intelligent Customer Service
> --
> 
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> 856-380-0894 x5000
> 
> 
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RE : [asterisk-users] Upgrade 4 to 8 Analog Lines Question

[f6hqz-m]

Or a new Digium TDM880B replacing the old TDM40B for only one IRQ...

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jim Freeze
Envoyé : lundi 9 avril 2007 15:15
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Upgrade 4 to 8 Analog Lines Question


Hello

I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru
an adtran board. I want to add 4 more analog lines. Currently I have a
Digium TDM40B. I'm wondering what the best upgrade path is, where I define
'best' as the solution that is most likely to work without problems (like
interupt conflicts) and work with my current echo tuning .

I see my purchase options as follows:

1) TDM40B - use with the current TDM40B
2) Sangoma Remora A20200 - use with the current TDM40B
3) Sangoma Remora A20400 - replace the current TDM40B


Any info will be greatly appreciated.

Thanks

Jim


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Re: [asterisk-users] sip_header=value?

[Rizwan Hisham]

I have tried it, it doesnt work

On 4/9/07, Hermann Wecke <[EMAIL PROTECTED]> wrote:


Rizwan Hisham wrote:
> is there anyway i can set SIP_HEADER(To) to the value i like?

If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Rizwan Hisham
Software Engineer
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RE: [asterisk-users] Verizon-Vonage Lawsuit

[Salvatore Giudice]

I think it's a small, feather covered appendage.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (702) 979-2906
Fax: (212) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Monday, April 09, 2007 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit

Christopher Chan wrote:
> 
> Welcome to China. Most Hong Kong'ers loathe mainland chinese and if they 
> could, they would never get a job in China. I get plenty of hack 
> attempts too from China however I doubt that is due to the same 
> sentiment in China.
> 
> If you want to find someone to blame, please look no further than the US 
> where your chicken boners are in league with crackers and virus writers 
> to create botnets to send their spam. This is of course besides the 
> ignorance of those who own computers in China (man, computers there are 
> infested with virii, worms and trojans) that run that most secure of 
> operating systems Microsoft Windows and those who actually get paid by 
> chicken boners to host their crap.
> 
> Oh, there are plenty of hack attempts from Korea too. Are you going to 
> add Korea to the list of 'IP' violators too?


Just curious,

Christopher, what is a chicken boner?


-- 

Warm Regards,

Lee


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RE: [asterisk-users] Upgrade 4 to 8 Analog Lines Question

[Michelle Dupuis]

Look at a channel bank with MGCP/TDMoE/USB interface (there are plenty
around).  We too prefer to keep fxs/fxo hardware outside of the * box.

Michelle Dupuis
Technical Support Specialist

Generation Software - Linux and Asterisk solutions and support.  Visit us at
www.generationd.com
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze
Sent: Monday, April 09, 2007 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Upgrade 4 to 8 Analog Lines Question

Hello

I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru
an adtran board. I want to add 4 more analog lines.
Currently I have a Digium TDM40B. I'm wondering what the best upgrade path
is, where I define 'best' as the solution that is most likely to work
without problems (like interupt conflicts) and work with my current echo
tuning .

I see my purchase options as follows:

1) TDM40B - use with the current TDM40B
2) Sangoma Remora A20200 - use with the current TDM40B
3) Sangoma Remora A20400 - replace the current TDM40B


Any info will be greatly appreciated.

Thanks

Jim


--
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Re: [asterisk-users] Verizon-Vonage Lawsuit

[Lee Jenkins]

Christopher Chan wrote:


Welcome to China. Most Hong Kong'ers loathe mainland chinese and if they 
could, they would never get a job in China. I get plenty of hack 
attempts too from China however I doubt that is due to the same 
sentiment in China.


If you want to find someone to blame, please look no further than the US 
where your chicken boners are in league with crackers and virus writers 
to create botnets to send their spam. This is of course besides the 
ignorance of those who own computers in China (man, computers there are 
infested with virii, worms and trojans) that run that most secure of 
operating systems Microsoft Windows and those who actually get paid by 
chicken boners to host their crap.


Oh, there are plenty of hack attempts from Korea too. Are you going to 
add Korea to the list of 'IP' violators too?



Just curious,

Christopher, what is a chicken boner?


--

Warm Regards,

Lee


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[asterisk-users] Upgrade 4 to 8 Analog Lines Question

[Jim Freeze]

Hello

I have an office with a T1 that provides 4 (out of 8) analog PSTN
lines thru an adtran board. I want to add 4 more analog lines.
Currently I have a Digium TDM40B. I'm wondering what the best
upgrade path is, where I define 'best' as the solution that
is most likely to work without problems (like interupt conflicts)
and work with my current echo tuning .

I see my purchase options as follows:

1) TDM40B - use with the current TDM40B
2) Sangoma Remora A20200 - use with the current TDM40B
3) Sangoma Remora A20400 - replace the current TDM40B


Any info will be greatly appreciated.

Thanks

Jim


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Re: [asterisk-users] How well does a celldock work with Asterisk?

[Dave Fullerton]

Joe Acquisto wrote:

Steve Prior <[EMAIL PROTECTED]> Wrote: 4/6/2007 8:30 PM:

Steve Prior wrote:
I've seen in the wiki that it is possible to use a celldock device to 
use a cell phone as a PSTN line to Asterisk, but I haven't seen any 
comments as to how well this actually works.  I was thinking about 
hooking a celldock to a FXO input of my Digium TDM400P card and use it 
to connect via bluetooth to my RAZR V3C.  I am aware of the software 
solution (chan_bluetooth), but my Asterisk box is a bit far away from 
where I want to keep the phone so the celldock seems to be the more 
convenient solution for me.


Any comments about the sound quality or issues in making it work?
I just found out that the celldock I'm talking about is also called the 
Dock-N-Talk.


I look forward to hearing about experiences in using it with Asterisk.

Steve



My curiosity is aroused, as well.   I would want to use these to allow me to 
eliminate my POTS lines entirely, and go to cell service.   This, due to the 
poor quality (and comparatively high cost) of Verizon service in this area.
Seems I would still need at least one POTS line, for FAX machine.  Can't use 
IAX in any form due to highly asymetrical link at this location.

joe a



I have a Dock-n-Talk at home I use to connect my motorola V60i via a 
cable so I can't comment on bluetooth. I needed it because for some 
reason I can only get good cell reception in my bedroom. It works well 
enough. You can certainly tell you are talking over a cell connection 
and not a POTS line (it's a little noisier) and now and then I have 
slight echo. Caller ID is passed through (number only) as well.

All in all I'm happy with the purchase.

-Dave
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[asterisk-users] Asterisk installation issue - CLI showing 0 active channels

[Vijay Gaur]

Hi All,
  I would appreciate a lot if you could help me. I have installed Asterisk
1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have also
installed 1 FXO port card: Digium TDM400P.
After loading zaptel driver I could see my digium card's led glow green.
Tested with zttool that its in OK state. I have configured fxsks=4 in
zaptel.conf(channel 4 cause FXO module is on port 4). I have also configured
channel 4 in incoming context in zapata.conf.

After all configuration I started asterisk with vgc and made a call, but
I didn't see any call log on asterisk console.
When I did core show channels its showing 0 active channels. Looks like I
did something wrong in my config.

Also I dont see red-alarm in zttool if I take out pstn line.

I have taken pstn line from my Vonage adapter and connecting that to my
digium card.

Any help will be appreciated.
Regards,
Vijay Gaur
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Re: [asterisk-users] sip_header=value?

[Hermann Wecke]

Rizwan Hisham wrote:

is there anyway i can set SIP_HEADER(To) to the value i like?


If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Re: [asterisk-users] How well does a celldock work with Asterisk?

[Joe Acquisto]

Steve Prior <[EMAIL PROTECTED]> Wrote: 4/6/2007 8:30 PM:
> Steve Prior wrote:
>> I've seen in the wiki that it is possible to use a celldock device to 
>> use a cell phone as a PSTN line to Asterisk, but I haven't seen any 
>> comments as to how well this actually works.  I was thinking about 
>> hooking a celldock to a FXO input of my Digium TDM400P card and use it 
>> to connect via bluetooth to my RAZR V3C.  I am aware of the software 
>> solution (chan_bluetooth), but my Asterisk box is a bit far away from 
>> where I want to keep the phone so the celldock seems to be the more 
>> convenient solution for me.
>> 
>> Any comments about the sound quality or issues in making it work?
> 
> I just found out that the celldock I'm talking about is also called the 
> Dock-N-Talk.
> 
> I look forward to hearing about experiences in using it with Asterisk.
> 
> Steve
> 

My curiosity is aroused, as well.   I would want to use these to allow me to 
eliminate my POTS lines entirely, and go to cell service.   This, due to the 
poor quality (and comparatively high cost) of Verizon service in this area.
Seems I would still need at least one POTS line, for FAX machine.  Can't use 
IAX in any form due to highly asymetrical link at this location.

joe a

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Re: [asterisk-users] sip_header=value?

[Rizwan Hisham]

No, The problem is, When i dial like this:

Dial(SIP/[EMAIL PROTECTED])
The "To" header field received on the peer asterisk contains the extension
which i dialed. and whenevr i dial like this:

Dial(SIP/user)
The "To" Header field received on peer asterisk contains the s extension
instead of the dialed extension for the user.

I want it to set it to the dialed extension always. So is it possible in the
dialplan or thru agi script or thru anyother means, or do i have to jump
into the source code?

On 4/9/07, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:



Something like
exten => s,1,SetVar(ALERT_INFO=something)



--
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Int:  (305) 704-7249  Fax: (815)301-9759 
 UK 44.207.183.0271 
Cell: 264-235-5670 
Yahoo IM: [EMAIL PROTECTED]


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Rizwan Hisham
Software Engineer
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Re: [asterisk-users] sip_header=value?

[Chris Mason (Lists)]

Something like
exten => s,1,SetVar(ALERT_INFO=something)



--
Chris Mason
(264) 497-5670  Fax: (264) 497-8463 

Int:  (305) 704-7249  Fax: (815)301-9759 
 UK 44.207.183.0271 
Cell: 264-235-5670 
Yahoo IM: [EMAIL PROTECTED] 



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[asterisk-users] sip_header=value?

[Rizwan Hisham]

Hi all,
is there anyway i can set SIP_HEADER(To) to the value i like?

--
Regards
Rizwan Hisham
Software Engineer
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Re: [asterisk-users] chan-capi-HEAD and Asterisk 1.4.2

[Armin Schindler]

On Mon, 9 Apr 2007, Peer Oliver Schmidt wrote:
> Hello Armin (and happy easter),
> 
> thanks for you continuing support.
> 
> > Can you please try HEAD version of SVN trunk (443)?
> 
> Did checkout the 443.
> 
> It works without any verbosity.
> 
> THANK YOU! I'll buy you a beer, if you ever happen to come to the
> northern part of Germany.

Thank you, but real thanks should go to the bug reporter of PR#28
on bugs.melware.net.

Armin

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