date:20080108

2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]:

 Olivier wrote:

  Is there way for an Asterisk server to check if a sip phone is forwarded
  without bothering phone's user ?

 No.

  I was thinking of some Alert-Info option that would let the phone reply
  with a 302 Moved Temporarily or 182 Queued message and not let the phone
  ring or display anything on its screen.

 According to the SIP RFC, a SIP endpoint is supposed to respond to an
 OPTIONS message the same way that it would respond to an INVITE message
 with the identical destination, but I've never seen a phone respond to
 an OPTIONS message with anything but '200 OK', even when a redirect
 (forward) is in place.


So, the alternative option is to play with html and use phone embedded html
server to get this redirection data.

Cheers

--
 Kevin P. Fleming
 Director of Software Technologies
 Digium, Inc. - The Genuine Asterisk Experience (TM)


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Re: [asterisk-users] app_rxfax.c and app_txxfax.c where?

2008-01-08 Thread Jonn R Taylor

http://www.taylortelephone.com/asterisk/





  _

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, 
David C
Sent: Monday, January 07, 2008 11:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] app_rxfax.c and app_txxfax.c where?



Hi All,



Where can I find copies of the app_rxfax.c, app_txfax.c and 
apps_Makefile.patch.  They don't seem to be located at soft-switch.org anymore.



I am currently trying to compile Asterisk 1.2.26.1 and need the fax components.



Thanks.



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Re: [asterisk-users] Limiting number of simultaneous calls in E1 line

2008-01-08 Thread Christian Victor


 I have a standard E1 line, but want to receive only 10 calls
 simultaneously. I want to give engaged tone to the 11th caller
 onwards. Can I configure E1 to do this?


Yes - that can be done on the carrier side. Lines can be configured to be
outgoing or incoming only.

Christian
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Re: [asterisk-users] Background Noise Elimination

2008-01-08 Thread Jerry Jones


On Jan 7, 2008, at 6:19 PM, Matt Riddell wrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Norman Franke wrote:
 Greetings!

 We have a somewhat noisy background in our call center, and I'd  
 like to
 reduce this. Obviously, we could plaster the walls with sound  
 absorbing
 material, but is there anything we can do in software either using  
 any
 algorithms for our open source-based SIP library or inside Asterisk
 itself? Related to this, anyone have a good source for good panels?

 We are using Plantronics noise canceling headsets, which don't really
 seem to work all that well. Our ancient system handled noise much
 better, but I suspect that was partly due to the Dialogic ADPCM
 algorithm used that just reduced the intelligibility of lower volume
 noises in general. We are using PCMU direct from the agent's mic to
 through Asterisk to PRIs, so we don't suffer from compression
 artifacts. The down side, is that you can make out even very quiet
 conversations in the background (like 3 agents to one side.)

 How have people handled this? I'm experimenting with a noise gate  
 that
 will lower the volume when the agent isn't talking, but that won't  
 help
 when the agent is talking.

 Nah, there's nothing really.

 The noise gate is your best bet.  I would assume that while an  
 agent is
 talking the customer will be listening to the agent, so the background
 noise will hardly be noticeable.

 The issue is, while two people are talking its pretty hard to remove
 just one of them from a wave file.

 Try the noise gate and see how you go.

 Oh, you might want to try a downwards expander instead (a noise  
 gate but
 with ratio as well as threshold).


We have an IP600 located in our colo, a very noisy environment. For a  
spooky experience make a phone call and pass the call through a  
Ditech audio processor in the path of the PRIs. You will hear no  
background noise. You can even use the speakerphone. Even Polycom to  
Polycom is not too bad. But an all IP path to anything else and you  
cant hardly hear the other person.



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Re: [asterisk-users] disable call waiting by default

2008-01-08 Thread Yehavi Bourvine +972-8-9489444

 I've connected some analogic phone to some fxs modules on an analogic card.

 I want to disable by default the call waiting sound.

 I know that dialing *70 before to call the call waiting is disabled
 until the next call, but isn't there a setting or a dialplan command
 to set up this automatically?

If you mean that there is no waiting call then use DEV_STATE function to see
whether the called extension is in a call; if so - call the Busy() application.
If you need more details then search in the lists's history - I;ve posted a
code fragment about  a month ago.

 __yehavi:

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[asterisk-users] Bugs??

2008-01-08 Thread Abdul

Good Day All,

I am facing a serious problem since I started to use asterisk. I dont know if 
it is a bug or some one already solved this.

Currently I am running 120 VoIP SIP channels on my asterisk server but each day 
2, 3 calls got hanged in asterisk,  and on asterisk CLI show channels showing 
us as call UP but in real there is no call.

When asterisk restarted the hanged calls removed from CLI with very high 
duration which damaged our billing system and customers accounts goes in high 
negative.

First I tried to get call info from asterisk mysql CDR using billsec field but 
the same result then I create PERL AGI to get duration from ANSWEREDTIME and 
same result.

Please have a look to the following URL which I put the result of show channel 
channelname you can see the DIALSTATUS=CONGESTION but Elapsed Time: 
20h54m16s which really strange and out of my mind to control such as call.

http://www.emafone.net/bugs.html

Please advice us if it is Bug and solved in some ver or its need some 
configuration to avoid this issue.

This is in both ver of asterisk 1.2 and 1.4




 
Regard,
   
-
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Re: [asterisk-users] Bugs??

2008-01-08 Thread Mike Trest - Personal

When similar problem occurred, I traced the issue to remote GSM
gateway with poor protocol stack.

The asterisk was doing exactly what it was supposed to do.

The IMMEDIATE work around we used was to put maximum call timer into
extensions.conf

exten = s, 6,Set(TIMEOUT(absolute)=3660)

This gives one hour+one minute. With average call duration below 30 minutes
this worked quite well for our GSM traffic purposes.
You set to any value appropriate to your traffic.
..mike..
Currently I am running 120 VoIP SIP channels on my asterisk server
but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI
show channels showing us as call UP but in real there is no call.

When asterisk restarted the hanged calls removed from CLI with very
high duration which damaged our billing system and customers
accounts goes in high negative.

First I tried to get call info from asterisk mysql CDR using billsec
field but the same result then I create PERL AGI to get duration
from ANSWEREDTIME and same result.

Please have a look to the following URL which I put the result of
show channel channelname you can see the DIALSTATUS=CONGESTION
but Elapsed Time: 20h54m16s which really strange and out of my mind
to control such as call.

http://www.emafone.net/bugs.html

Please advice us if it is Bug and solved in some ver or its need
some configuration to avoid this issue.

This is in both ver of asterisk 1.2 and 1.4

Regard,

Looking for last minute shopping deals?
http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shoppingFind
them fast with Yahoo! Search.

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Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

2008/1/8, Steve Langstaff [EMAIL PROTECTED]:

  That's going to be pretty phone-specific. How about asking your phone
 supplier to fix their phone so that it responds to OPTIONS correctly?


Yes, you're right but RFC3261 doesn't specify such 302 replies.
So I'm very pessimistic about my phone supplier answer.



--
 *From:* [EMAIL PROTECTED] [mailto:
 [EMAIL PROTECTED] *On Behalf Of *Olivier
 *Sent:* 08 January 2008 12:50
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] How to check if a SIP phone is
 forwardedwithout ringing it ?

 2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]:
 
  Olivier wrote:
 
   Is there way for an Asterisk server to check if a sip phone is
  forwarded
   without bothering phone's user ?
 
  No.
 
   I was thinking of some Alert-Info option that would let the phone
  reply
   with a 302 Moved Temporarily or 182 Queued message and not let the
  phone
   ring or display anything on its screen.
 
  According to the SIP RFC, a SIP endpoint is supposed to respond to an
  OPTIONS message the same way that it would respond to an INVITE message
  with the identical destination, but I've never seen a phone respond to
  an OPTIONS message with anything but '200 OK', even when a redirect
  (forward) is in place.


 So, the alternative option is to play with html and use phone embedded
 html server to get this redirection data.

 Cheers

 --
  Kevin P. Fleming
  Director of Software Technologies
  Digium, Inc. - The Genuine Asterisk Experience (TM)
 
 

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Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

2008-01-08 Thread Steve Langstaff

Section 11.2 of RFC 3261 details the Processing of OPTIONS Request
 
   The response to an OPTIONS is constructed using the standard rules
   for a SIP response as discussed in Section 8.2.6.  The response code
   chosen MUST be the same that would have been chosen had the request
   been an INVITE.  That is, a 200 (OK) would be returned if the UAS is
   ready to accept a call, a 486 (Busy Here) would be returned if the
   UAS is busy, etc.  This allows an OPTIONS request to be used to
   determine the basic state of a UAS, which can be an indication of
   whether the UAS will accept an INVITE request.
 
Section 21.3.3 of RFC3261 details the 302 Moved Temporarily response
code.
 
Looks to me like those two things should interwork just fine.




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: 08 January 2008 14:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to check if a SIP phone
isforwardedwithout ringing it ?




2008/1/8, Steve Langstaff [EMAIL PROTECTED]: 

That's going to be pretty phone-specific. How about
asking your phone supplier to fix their phone so that it responds to
OPTIONS correctly?


Yes, you're right but RFC3261 doesn't specify such 302 replies.
So I'm very pessimistic about my phone supplier answer.


 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: 08 January 2008 12:50
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] How to check if a
SIP phone is forwardedwithout ringing it ?



2008/1/7, Kevin P. Fleming
[EMAIL PROTECTED]: 


Olivier wrote:

 Is there way for an Asterisk server to
check if a sip phone is forwarded
 without bothering phone's user ?

No.

 I was thinking of some Alert-Info
option that would let the phone reply 
 with a 302 Moved Temporarily or 182
Queued message and not let the phone
 ring or display anything on its
screen.

According to the SIP RFC, a SIP endpoint
is supposed to respond to an
OPTIONS message the same way that it
would respond to an INVITE message 
with the identical destination, but I've
never seen a phone respond to
an OPTIONS message with anything but
'200 OK', even when a redirect
(forward) is in place.


So, the alternative option is to play with html
and use phone embedded html server to get this redirection data. 

Cheers



--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk
Experience (TM) 





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Re: [asterisk-users] disable call waiting by default

2008-01-08 Thread Don Pobanz

From: nik600 on Tuesday, January 08, 2008 6:02 AM
 
 I've connected some analogic phone to some fxs modules on an 
 analogic card.
 
 I want to disable by default the call waiting sound.

In zapata.conf 
Callwaiting = no

Don Pobanz

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Re: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help

nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when 
using meetme conferencing... I suggest you spend time elsewhere in * 
until you get a digium tdm400 w/ or w/o any daughter modules...  you 
just need the board for the timing device you don't actually need any 
modules. $195 for tdm400p + one mondule.. developers kit...
daveC

Nhadie wrote:
 hi shane,

 thanks for your reply. i actually tried 3 phones dialled to the 
 conference, but cant here anything from those phones. i also enabled the 
 usercount so i can hear something at least. but still no sound.
 i'm using ztdummy, as i dont have a card yet.

 regards,
 nhadie

 Shane D wrote:
   
 Wouldn't you need someone besides yourself in the conference?

 On 1/7/08, Nhadie [EMAIL PROTECTED] wrote:
 
 Hi All,

 kind of need help on the conference module, i'm using freepbx and
 enabled conferencing, i created a conference number, 6000. when i dial
 to it, my phone says it is connected but i'm hearing nothing, maybe logs
 below can help you.

 also, when i hang up the phone, the conference did not disconnect me.
 how can i end a conference? thank you

  -- Executing Macro(SIP/104-519e, user-callerid|) in new stack
  -- Executing NoOp(SIP/104-519e, user-callerid: device 104) in
 new stack
  -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?report) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?start) in new stack
  -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack
  -- Executing NoOp(SIP/104-519e, REALCALLERIDNUM is 104) in new
 stack
  -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack
  -- Executing Set(SIP/104-519e, AMPUSERCIDNAME=104) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?report) in new stack
  -- Executing Set(SIP/104-519e, AMPUSERCID=104) in new stack
  -- Executing Set(SIP/104-519e, CALLERID(all)=104 104) in
 new stack
  -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack
  -- Executing NoOp(SIP/104-519e, TTL:  ARG1: ) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?continue) in new stack
  -- Executing Set(SIP/104-519e, __TTL=64) in new stack
  -- Executing GotoIf(SIP/104-519e, 1?continue) in new stack
  -- Goto (macro-user-callerid,s,23)
  -- Executing NoOp(SIP/104-519e, Using CallerID 104 104) in
 new stack
  -- Executing Set(SIP/104-519e, MEETME_ROOMNUM=6000) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?USER) in new stack
  -- Executing Answer(SIP/104-519e, ) in new stack
  -- Executing Wait(SIP/104-519e, 1) in new stack
  -- Executing Set(SIP/104-519e, MEETME_OPTS=) in new stack
  -- Executing Goto(SIP/104-519e, STARTMEETME|1) in new stack
  -- Goto (from-internal,STARTMEETME,1)
  -- Executing MeetMe(SIP/104-519e, 6000||) in new stack


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I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
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Re: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help

2008-01-08 Thread Steve Totaro

Is this also the case with FC7?  I have heard multiple times that FC7 has a
different/better timing method.  I wonder if this will help with ztdummy.

Thanks,
Steve Totaro

On 1/8/08, dave cantera [EMAIL PROTECTED] wrote:

 nhadie,
 meetme requires a zaptel timing device... ztdummy is unreliable when
 using meetme conferencing... I suggest you spend time elsewhere in *
 until you get a digium tdm400 w/ or w/o any daughter modules...  you
 just need the board for the timing device you don't actually need any
 modules. $195 for tdm400p + one mondule.. developers kit...
 daveC

 Nhadie wrote:
  hi shane,
 
  thanks for your reply. i actually tried 3 phones dialled to the
  conference, but cant here anything from those phones. i also enabled the
  usercount so i can hear something at least. but still no sound.
  i'm using ztdummy, as i dont have a card yet.
 
  regards,
  nhadie
 
  Shane D wrote:
 
  Wouldn't you need someone besides yourself in the conference?
 
  On 1/7/08, Nhadie [EMAIL PROTECTED] wrote:
 
  Hi All,
 
  kind of need help on the conference module, i'm using freepbx and
  enabled conferencing, i created a conference number, 6000. when i dial
  to it, my phone says it is connected but i'm hearing nothing, maybe
 logs
  below can help you.
 
  also, when i hang up the phone, the conference did not disconnect me.
  how can i end a conference? thank you
 
   -- Executing Macro(SIP/104-519e, user-callerid|) in new stack
   -- Executing NoOp(SIP/104-519e, user-callerid: device 104) in
  new stack
   -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack
   -- Executing GotoIf(SIP/104-519e, 0?report) in new stack
   -- Executing GotoIf(SIP/104-519e, 0?start) in new stack
   -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new
 stack
   -- Executing NoOp(SIP/104-519e, REALCALLERIDNUM is 104) in
 new
  stack
   -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack
   -- Executing Set(SIP/104-519e, AMPUSERCIDNAME=104) in new
 stack
   -- Executing GotoIf(SIP/104-519e, 0?report) in new stack
   -- Executing Set(SIP/104-519e, AMPUSERCID=104) in new stack
   -- Executing Set(SIP/104-519e, CALLERID(all)=104 104) in
  new stack
   -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new
 stack
   -- Executing NoOp(SIP/104-519e, TTL:  ARG1: ) in new stack
   -- Executing GotoIf(SIP/104-519e, 0?continue) in new stack
   -- Executing Set(SIP/104-519e, __TTL=64) in new stack
   -- Executing GotoIf(SIP/104-519e, 1?continue) in new stack
   -- Goto (macro-user-callerid,s,23)
   -- Executing NoOp(SIP/104-519e, Using CallerID 104 104)
 in
  new stack
   -- Executing Set(SIP/104-519e, MEETME_ROOMNUM=6000) in new
 stack
   -- Executing GotoIf(SIP/104-519e, 0?USER) in new stack
   -- Executing Answer(SIP/104-519e, ) in new stack
   -- Executing Wait(SIP/104-519e, 1) in new stack
   -- Executing Set(SIP/104-519e, MEETME_OPTS=) in new stack
   -- Executing Goto(SIP/104-519e, STARTMEETME|1) in new stack
   -- Goto (from-internal,STARTMEETME,1)
   -- Executing MeetMe(SIP/104-519e, 6000||) in new stack
 
 
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 --
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 I should buy her a Videophone2008!

 Truly, The Next Best Thing to Being There!
 --

 WorldWideVideoPhones.com
 856.380.0894




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Re: [asterisk-users] Early media support for Asterisk behind NAT


8 jan 2008 kl. 07.41 skrev Mayur:

 Hi,
I have asterisk 1.4.16 behind a NAT-FW which is using a hosted  
 SIP trunk for PSTN calling. Asterisk is configured to support nat  
 with nat=yes in sip.conf. Now the hosted PSTN Gateway supports  
 symmetric RTP and early media using 183 Session Progress. So If I  
 call a PSTN number which has IVR message played before the call is  
 connected (via 183), those media RTP packets do not reach the  
 asterisk inside till asterisk sends out media packet to the PSTN  
 gateway. I have used rtpkeepalive option and set it to 1 sec. But it  
 seems that I drop rtp voice packets in the initial instructions  
 played by the IVR.

 How do I make sure that asterisk sends RTP packets (null rtp) to the  
 PSTN gateway just after receiving the media details in 183 SDP to  
 open the firewall port from inside?

That's a very interesting question. We are able to receive media as  
soon as we send the INVITE, but I am unsure on when we actually start  
sending media. Turn on RTP debugging in your asterisk to check. I  
would assume that if you have rtpkeepalive, we should start sending as  
soon as we get somewhere to send to, which in this case is when we get  
the SDP in the 183. There might be issues with some packets being sent  
at the same time as the gateway sends 183. With QoS priority for  
media, these may arrive to the NAT before the 183 SIP reply, which  
will be a problem in this NATted situation.

There's no way we can actually send anything before the 183, so there  
will always be time between SDP exchange and first functional media  
packet in NAT situations. I always consider this when playing prompts  
and wait at least a second before important audio begins.

/O

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Re: [asterisk-users] GotoIf() help





glenn,
what an interesting way to use GotoIf() and 9. didn't know you
could do that in GotoIf()!
you could have used (broken out) the individual services 
[trunklocal]
[trunkld]
[trunktollfree]
and just included the above individual context in with the groups that
you allowed a particular class of service to ...
daveC

Glenn Cobb wrote:

  
  
  Greetings all,
  
  I'm not real good with dial
plan programming and need some help. I've looked at the 2nd edition of
the Asterisk book about GotoIf()and have a basic idea what I need to
do but not sure aboutthe correct way or the best way,to set itup. I
need to branch based on whether the dialed number is long distance
(international or not) or not. I have branch offices on SIP and IAX
trunks that have 4 digit extensions and one office has a 1000 range for
their extensions so I have to make sure I don't pick that up as dialing
long distance.I think what I have below will workbut it can probably
be cleaned up alot. Any help is greatly appreciated.
  
  
  exten =
s,n,GotoIf($[${DIAL_NUMBER} = 011. ] ? yescode : steptwo)
  
  exten =
s,n,(steptwo),GotoIf($[${DIAL_NUMBER} = 9XX. ] ? yescode :
stepthree)
  
  exten =
s,n,(stepthree),GotoIf($[${DIAL_NUMBER} = 1NXXNX. ] ? yescode : nocode)
  
  exten = s,n,(yescode),Playback(please-enter-theaccounting)
  exten = s,n,Read(account|number|8)
  exten = s,n,SetAccount(${account})
  exten =
s,n,(nocode),Blah, Blah
  
  
  Thanks,
  
  Glenn
  

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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.516 / Virus Database: 269.17.13/1209 - Release Date: 01/04/2008 12:05 PM
  


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[asterisk-users] :POSSIBLE SPAM: Re: conferencing help

2008-01-08 Thread Nhadie

hi dave thank you for the reply. i have loaded zap and using only 
ztdummy but still can't hear anything when i dial ti my conference, i 
think this explains it already. will a sangoma card do?

dave cantera wrote:
 nhadie,
 meetme requires a zaptel timing device... ztdummy is unreliable when 
 using meetme conferencing... I suggest you spend time elsewhere in * 
 until you get a digium tdm400 w/ or w/o any daughter modules...  you 
 just need the board for the timing device you don't actually need any 
 modules. $195 for tdm400p + one mondule.. developers kit...
 daveC
 
 Nhadie wrote:
 hi shane,

 thanks for your reply. i actually tried 3 phones dialled to the 
 conference, but cant here anything from those phones. i also enabled the 
 usercount so i can hear something at least. but still no sound.
 i'm using ztdummy, as i dont have a card yet.

 regards,
 nhadie

 Shane D wrote:
   
 Wouldn't you need someone besides yourself in the conference?

 On 1/7/08, Nhadie [EMAIL PROTECTED] wrote:
 
 Hi All,

 kind of need help on the conference module, i'm using freepbx and
 enabled conferencing, i created a conference number, 6000. when i dial
 to it, my phone says it is connected but i'm hearing nothing, maybe logs
 below can help you.

 also, when i hang up the phone, the conference did not disconnect me.
 how can i end a conference? thank you

  -- Executing Macro(SIP/104-519e, user-callerid|) in new stack
  -- Executing NoOp(SIP/104-519e, user-callerid: device 104) in
 new stack
  -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?report) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?start) in new stack
  -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack
  -- Executing NoOp(SIP/104-519e, REALCALLERIDNUM is 104) in new
 stack
  -- Executing Set(SIP/104-519e, AMPUSER=104) in new stack
  -- Executing Set(SIP/104-519e, AMPUSERCIDNAME=104) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?report) in new stack
  -- Executing Set(SIP/104-519e, AMPUSERCID=104) in new stack
  -- Executing Set(SIP/104-519e, CALLERID(all)=104 104) in
 new stack
  -- Executing Set(SIP/104-519e, REALCALLERIDNUM=104) in new stack
  -- Executing NoOp(SIP/104-519e, TTL:  ARG1: ) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?continue) in new stack
  -- Executing Set(SIP/104-519e, __TTL=64) in new stack
  -- Executing GotoIf(SIP/104-519e, 1?continue) in new stack
  -- Goto (macro-user-callerid,s,23)
  -- Executing NoOp(SIP/104-519e, Using CallerID 104 104) in
 new stack
  -- Executing Set(SIP/104-519e, MEETME_ROOMNUM=6000) in new stack
  -- Executing GotoIf(SIP/104-519e, 0?USER) in new stack
  -- Executing Answer(SIP/104-519e, ) in new stack
  -- Executing Wait(SIP/104-519e, 1) in new stack
  -- Executing Set(SIP/104-519e, MEETME_OPTS=) in new stack
  -- Executing Goto(SIP/104-519e, STARTMEETME|1) in new stack
  -- Goto (from-internal,STARTMEETME,1)
  -- Executing MeetMe(SIP/104-519e, 6000||) in new stack


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Re: [asterisk-users] :POSSIBLE SPAM: Re: conferencing help

 dave cantera wrote:
 nhadie,
 meetme requires a zaptel timing device... ztdummy is unreliable when
 using meetme conferencing.

On Wed, 9 Jan 2008, Nhadie wrote:

 hi dave thank you for the reply. i have loaded zap and using only
 ztdummy but still can't hear anything when i dial ti my conference, i
 think this explains it already. will a sangoma card do?

I use ztdummy with meetme conferencing and it works fine on CentOS 4.5. 
Ztdummy is not an issue until you get xx callers in xx conferences.

I think (but have no empirical data to back it up) that a card yields 
better sound quality at higher call levels.

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] Lamps on Snom phones

2008-01-08 Thread Phil Knighton

Sorry for slow response, been away.

Stefan, thankyou. I've made the changes you suggested to my sip.conf -
and all is back to normal.

Thanks to everyone else for your suggestions.

Phil 

-Original Message-
From: Stefan Guenther [mailto:[EMAIL PROTECTED] 
Sent: 03 January 2008 16:28
To: Phil Knighton
Cc: asterisk-users@lists.digium.com
Subject: Re: Re: Lamps on Snom phones

Hello Phil,

please check the following details in your asterisk configuration and on
your phones. These are the settings that work for me:

sip.conf

[general]
limitonpeers=yes
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
useclientcode=yes
canreinvite=yes

[user1]
secret=user1
host=dynamic
username=user1
callerid=user1 97
dtmfmode=rfc2833
context=local
type=friend
callgroup=1
pickupgroup=1
qualify=yes
vmexten=80297
call-limit=20
subscribecontext=local

extensions.conf

exten = 97,hint,SIP/user1
exten = 98,hint,SIP/smguenther

On the SNOM phones:

Support broken Registrar: ON
Use user:phone: OFF
Filter Packets from Registrar: OFF

Function Key P6:
ACTIVE / EXTENSION / sip:[EMAIL PROTECTED]

Hope that helps,

Stefan

-- 


in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de

 Schulungen  Installationen  
 Beratung   Support
  Voice-over-IP-Loesungen



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[asterisk-users] [Zaptel] Checking that TDM card works?

2008-01-08 Thread Vincent

Hello

Since TDM cards are known for being particular when it comes
to motherboards (PCI 2.2, etc.), I was wondering if there is a utility
that can check that the Zaptel driver works OK and can tell if the TDM
card is compatible?

That way, if an FXO module is not reporting an incoming call, we'd
know it's because of the Zaptel driver, and not something elsewhere.

Are dmesg, lspci -v, ztcfg -vv and zttool the only tools
available to investigate this issue?

Thank you.
==

PS: I'm using an OpenVox clone of the Digium card, with just one FXO
module. With the FXO module installed on plug #1, here's what I tried
so far. Note that ztcfg -vv says 1 channels to configure., while
Digium cards apparently say 1 channels configured:

= # dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.7
Zaptel Echo Canceller: MG2
ACPI: PCI Interrupt :00:0f.0[A] - Link [LNKD] - GSI 12 (level,
low) - IRQ 12
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXO (FCC mode)
Module 1: Not installed
Module 2: Not installed
Module 3: Not installed
Found a Wildcard TDM: Wildcard TDM400P REV E/F (1 modules)
[...]
usbcore: registered new driver wcusb
Wildcard USB FXS Interface driver registered
Registered tone zone 2 (France)

= # lspci -v

00:0f.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device b100:0003
Flags: bus master, medium devsel, latency 64, IRQ 12
I/O ports at c400 [size=256]
Memory at dfffe000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2

= # cat /etc/zaptel.conf
fxsks=1
loadzone=fr
defaultzone=fr

= # ztcfg -

Zaptel Version: 1.4.7
Echo Canceller: MG2
Configuration

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels to configure.

= # cat zapata.conf
[channels]
language=fr
context=my-phones
usecallerid=yes
hidecallerid=no
immediate=no

signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
channel=1

= # cat /etc/asterisk/extensions.conf
[general]

[globals]

[my-phones]
exten = s,1,Verbose(yes!)

= # service zaptel restart
Unloading zaptel hardware drivers:.
Loading zaptel framework:  [  OK  ]
Waiting for zap to come online...OK
Loading zaptel hardware modules: tor2.
 wct4xxp.
 wcte12xp.
 wct1xxp.
 wcte11xp.
 wctdm24xxp.
 wcfxo.
 wctdm.
 wcusb.
Running ztcfg:  [  OK  ]

= # service asterisk restart
Shutting down asterisk: Asterisk ended with exit status 0
Asterisk shutdown normally.
[  OK  ]
Starting asterisk: [  OK  ]

= # asterisk -vvr
asterisk*CLI 

=
Here, I call into the OpenVox card from a cellphone, but nothing is
shown in the Asterisk console :-/


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Re: [asterisk-users] Bugs??

2008-01-08 Thread Michiel van Baak

 Currently I am running 120 VoIP SIP channels on my asterisk server 
 but each day 2, 3 calls got hanged in asterisk,  and on asterisk CLI 
 show channels showing us as call UP but in real there is no call.
 
 When asterisk restarted the hanged calls removed from CLI with very 
 high duration which damaged our billing system and customers 
 accounts goes in high negative.
 
 First I tried to get call info from asterisk mysql CDR using billsec 
 field but the same result then I create PERL AGI to get duration 
 from ANSWEREDTIME and same result.
 
 Please have a look to the following URL which I put the result of 
 show channel channelname you can see the DIALSTATUS=CONGESTION 
 but Elapsed Time: 20h54m16s which really strange and out of my mind 
 to control such as call.
 
 http://www.emafone.net/bugs.html
 
 Please advice us if it is Bug and solved in some ver or its need 
 some configuration to avoid this issue.
 
 This is in both ver of asterisk 1.2 and 1.4

I guess you have a call to the Congestion dialplan function.
We found out in some locations this will go on forever.
A simple fix is to give the Congestion function call a max
duration like: exten = foo,n,Congestion(3) followed by a
Hangup()

-- 

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

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


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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

2008-01-08 Thread Tzafrir Cohen

On Tue, Jan 08, 2008 at 07:06:17PM +0100, Vincent wrote:
 Hello
 
   Since TDM cards are known for being particular when it comes
 to motherboards (PCI 2.2, etc.), I was wondering if there is a utility
 that can check that the Zaptel driver works OK and can tell if the TDM
 card is compatible?
 
 That way, if an FXO module is not reporting an incoming call, we'd
 know it's because of the Zaptel driver, and not something elsewhere.
 
 Are dmesg, lspci -v, ztcfg -vv and zttool the only tools
 available to investigate this issue?
 
 Thank you.
 ==
 
 PS: I'm using an OpenVox clone of the Digium card, with just one FXO
 module. With the FXO module installed on plug #1, here's what I tried
 so far. Note that ztcfg -vv says 1 channels to configure., while
 Digium cards apparently say 1 channels configured:

This change is simply due to different versions of Zaptel. Zaptel =
1.4.6 prints to configure because this message is printed (and has
always been prinetd) before the configuration is actually applied. And
hence fooled poor users into believing that their channels were properly
configured.

 = # cat zapata.conf
 [channels]
 language=fr
 context=my-phones
 usecallerid=yes
 hidecallerid=no
 immediate=no
 
 signalling=fxs_ks
 echocancel=yes
 echocancelwhenbridged=yes
 channel=1
 
 = # cat /etc/asterisk/extensions.conf
 [general]
 
 [globals]
 
 [my-phones]
 exten = s,1,Verbose(yes!)

In the Asterisk CLI run:

  core set verbose 3

And then see what happens when a call comes in.
Basically you miss an action to do after the Verbose line.

Alternatively, what is the output of:

cat /proc/zaptel/*
asterisk -rx 'zap show channels'

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

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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

2008-01-08 Thread Jared Smith

On Tue, 2008-01-08 at 19:06 +0100, Vincent wrote:
 Are dmesg, lspci -v, ztcfg -vv and zttool the only tools
 available to investigate this issue?

I always find that looking at the files that are generated
under /proc/zaptel is very enlightening as far as showing what the
zaptel drivers are seeing.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] What's the best ztdummy?

I have several servers using ztdummy as the timing source, some CentOS 
4.x, some CentOS 5.x, some Asterisk 1.2.x, some Asterisk 1.4.x.

zap show status differs between the servers:

ZTDUMMY/1 (source: Linux26) 1UNCONFIGUR 0  0  0
ZTDUMMY/1 (source: RTC) 1UNCONFIGUR 0  0  0
ZTDUMMY/1 1  UNCONFIGUR 0  0  0

Is one better than the other? What is the best timing source for ztdummy 
and what does its status look like?

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] Bugs??

2008-01-08 Thread Abdul

 We are not using any GSM Gateway for call carriers we have

Asterisk  TELES(iSWITCH) --- MCI

As Teles is world class telecoms product it should not make poor protocol stack.
In my AGI script already i am using 

TIMEOUT(absolute)to limit the call according to registrar balance.

I am thinking my be exten = foo,n,Congestion(3) function can solve
the issue but how i can call this i should call it after dial or before?

is (3) is max  Congestion time?

Thank You 




Abdul [EMAIL PROTECTED] wrote:
Abdul [EMAIL PROTECTED] wrote: Good Day All,

I am facing a serious problem since I started to use asterisk. I dont know if 
it is a bug or some one already solved this.

Currently I am running 120 VoIP SIP channels on my asterisk server but each day 
2, 3 calls got hanged in  asterisk,  and on asterisk CLI show channels 
showing us as call UP but in real there is no call.

When asterisk restarted the hanged calls removed from CLI with very high 
duration which damaged our billing system and customers accounts goes in high 
negative.

First I tried to get call info from asterisk mysql CDR using billsec field but 
the same result then I create PERL AGI to get duration from ANSWEREDTIME and 
same result.

Please have a look to the following URL which I put the result of show channel 
channelname you can see the DIALSTATUS=CONGESTION but Elapsed Time: 
20h54m16s which really strange and out of my mind to control such as  call.

http://www.emafone.net/bugs.html

Please advice us if it is Bug and solved in some ver or its need some 
configuration to avoid this issue.

This is in both ver of asterisk 1.2 and 1.4




  
Regard,

-
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-
Looking for last minute shopping deals?   Find them fast with Yahoo! Search.


 
Regard,
   
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Re: [asterisk-users] Distorted audio over Eicon Diva Server BRI

2008-01-08 Thread CSB


 Sounds very similar to an issue I was having.

 Are you using mISDN?

No. Incidentally, what's the benefit of using mISDN?

Regards

Cameron



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Re: [asterisk-users] BLF trouble

2008-01-08 Thread Lars Bensmann

On Fri, Jan 04, 2008 at 09:57:02PM +0100, Lars Bensmann wrote:
 I can see if I can install a vanilla 1.4 off-hours and just test the
 SIP-phones. Although I don't know when I will be able to do so.

OK. I tested this today it it behaved exactly like before. Hints work
for incoming calls but extensions are not marked as busy for outgoing
calls.

 Is there anybody who has the hints working for outgoing calls?

Does this really mean it just doesn't work? Nobody has working hints for
outgoing calls? I thought this should be a rather common setup.

Should I file a bug report for this?

Lars

-- 
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[asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

2008-01-08 Thread Jean-Louis curty

Hi,

I succesfully install spandsp chan_misdn and digium card. the rxfax works
fine and I get the fax result by email.
I would like to do the same using a Patton gw + zaptel but I can't receive
fax anymore,

the call comes in from ISDN in the Patton gw, patton sends it to asterisk,
asterisk run a macro to make a tif file using rxfax,
the tif file is correctly created but with a 0 size the call looks normal, 1
pages, 45 seconds and disconnect but the file is still 0,

anyone succeeded in this ?
Many (many) thanks!
jean-louis
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Re: [asterisk-users] Bugs??

2008-01-08 Thread Abdul


Sorry i forget to give my extentions config.

[clientsG]
exten = _x.,1,Set(UserN=${CALLERID(all)})
exten = _x.,2,Set(CalledNum=${EXTEN})
exten = _x.,3,Set(Stime=${DATETIME})
exten = _x.,4,Set(CID=${CALLERID})
exten = _x.,5,Set(HCA=${HANGUPCAUSE})
exten = _x.,6,Set(Cun=${UNIQUEID})
exten = _x.,7,AGI(routing.pl)
exten = h,1,DeadAGI(stop.pl)
exten = h,1,Hangup

routing.pl

$AGI-exec('Set',TIMEOUT(absolute)=$cstatus);
 my $dialstr = $gwtype/$gwip/$dialednum;
 $AGI-exec('Dial', $dialstr);#//Dial the number



Abdul [EMAIL PROTECTED] wrote:  We are not using any GSM Gateway for call 
carriers we have

Asterisk  TELES(iSWITCH) --- MCI

As Teles is world class telecoms product it should not make poor protocol stack.
In my AGI script already i am using 

TIMEOUT(absolute)to limit the call according to registrar balance.

I am thinking my be exten = foo,n,Congestion(3) function can solve
the issue but how i can call this i should call it after dial or before?

is (3) is max  Congestion time?

Thank You 




Abdul [EMAIL PROTECTED] wrote:
Abdul [EMAIL PROTECTED] wrote: Good Day All,

I am facing a serious problem since I started to use  asterisk. I dont know if 
it is a bug or some one already solved this.

Currently I am running 120 VoIP SIP channels on my asterisk server but each day 
2, 3 calls got hanged in  asterisk,  and on asterisk CLI show channels 
showing us as call UP but in real there is no call.

When asterisk restarted the hanged calls removed from CLI with very high 
duration which damaged our billing system and customers accounts goes in high 
negative.

First I tried to get call info from asterisk mysql CDR using billsec field but 
the same result then I create PERL AGI to get duration from ANSWEREDTIME and 
same result.

Please have a look to the following URL which I put the result of show channel 
channelname you can see the DIALSTATUS=CONGESTION but Elapsed Time: 
20h54m16s which really strange and out of my mind to control such as  call.

http://www.emafone.net/bugs.html

Please advice us if it is Bug and solved in some ver or its need  some 
configuration to avoid this issue.

This is in both ver of asterisk 1.2 and 1.4




  
Regard,

-
Looking for last minute shopping deals?   Find them fast with Yahoo! Search.


-
Looking for last minute shopping deals?   Find them fast with Yahoo! Search.


 
Regard,

-
Looking for last minute shopping deals?   Find them fast with Yahoo! Search.

   
-
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[asterisk-users] Simultaneous Callback?!

2008-01-08 Thread Douglas Garstang

We're doing callback here. Asterisk dials a number, waits for an answer, plays 
a prompt, dials a second number, and bridges the channels together.
Calls are initiated from the AMI.
No problems there. Easy stuff.

However, I'd like to know if it's possible to have Asterisk dial the same two 
numbers simultaneously, play the prompt to the first one that answers, dial the 
second one and bridge the two channels together.?

I'm not even sure how this would work within the limits of the dial plan. 
Normally, the dialing of the first leg is implicit (the channel in an AMI 
originate command), that is, there is no dial plan code for it, although you 
can specify a Local channel and asterisk will then jump into the dial plan to 
dial the first number. Once the first one answers, Asterisk jumps to the 
location specified by the second number (ie [EMAIL PROTECTED]) and calls it, 
and bridges them together.

How would this work with simultaneous numbers?









  

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Re: [asterisk-users] BLF trouble

2008-01-08 Thread Benny Amorsen

Lars Bensmann [EMAIL PROTECTED] writes:

 Does this really mean it just doesn't work? Nobody has working hints for
 outgoing calls? I thought this should be a rather common setup.

I would have imagined so too.

 Should I file a bug report for this?

I think it would be great if you did.


/Benny



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Re: [asterisk-users] Bugs??

On Tue, 8 Jan 2008, Abdul wrote:

 Sorry i forget to give my extentions config.

 [clientsG]
 exten = _x.,1,Set(UserN=${CALLERID(all)})
 exten = _x.,2,Set(CalledNum=${EXTEN})
 exten = _x.,3,Set(Stime=${DATETIME})
 exten = _x.,4,Set(CID=${CALLERID})
 exten = _x.,5,Set(HCA=${HANGUPCAUSE})
 exten = _x.,6,Set(Cun=${UNIQUEID})
 exten = _x.,7,AGI(routing.pl)
 exten = h,1,DeadAGI(stop.pl)
 exten = h,1,Hangup

Unrelated to your problem, but here's a suggestion -- use n instead of 
explicitly numbering your priorities. Like:

[clientsG]
exten = _x.,1,Set(UserN=${CALLERID(all)})
exten = _x.,n,Set(CalledNum=${EXTEN})
exten = _x.,n,Set(Stime=${DATETIME})
exten = _x.,n,Set(CID=${CALLERID})
exten = _x.,n,Set(HCA=${HANGUPCAUSE})
exten = _x.,n,Set(Cun=${UNIQUEID})
exten = _x.,n,AGI(routing.pl)
exten = h,1,DeadAGI(stop.pl)
exten = h,n,Hangup

Note that your hangup duplicated the h,1 priority.

Also, if you are setting these channel variables just so they can be 
picked up by your AGI, you can pass variables to the AGI application. 
Like:

AGI(routing.pl,${CALLERID(all)},${EXTEN})

or

AGI(routing.pl,--callerid=${CALLERID(all)},--exten=${EXTEN})

Since most of these are already being passed to your AGI in the AGI 
environment, passing them again on the command line would be a bit 
redundant.

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] BLF trouble


8 jan 2008 kl. 21.10 skrev Lars Bensmann:

 On Fri, Jan 04, 2008 at 09:57:02PM +0100, Lars Bensmann wrote:
 I can see if I can install a vanilla 1.4 off-hours and just test the
 SIP-phones. Although I don't know when I will be able to do so.

 OK. I tested this today it it behaved exactly like before. Hints work
 for incoming calls but extensions are not marked as busy for outgoing
 calls.

 Is there anybody who has the hints working for outgoing calls?

 Does this really mean it just doesn't work? Nobody has working hints  
 for
 outgoing calls? I thought this should be a rather common setup.

 Should I file a bug report for this?
Check the documentation in sip.conf...

There is a setting for enforcing the call limit for both inbound and  
outbound
on a peer only. That way you will have BLF for both directions on
a subscription.

/O

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Re: [asterisk-users] Bugs??

On Tue, 8 Jan 2008, Steve Edwards wrote:

 or

   AGI(routing.pl,--callerid=${CALLERID(all)},--exten=${EXTEN})

Oops -- assuming you use getopt_long() (or it's Perl equivalent).

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] Distorted audio over Eicon Diva Server BRI

2008-01-08 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Tim Panton wrote:
 On 8 Jan 2008, at 08:17, Armin Schindler wrote:
 
 On Tue, 8 Jan 2008, CSB wrote:
 We are experiencing slightly distorted audio with playing of  
 recordings on
 our Asterisk server when the call comes in over our Eicon Diva  
 Server BRI
 card. An example is an incoming call to IVR and playing some of the  
 standard
 Asterisk voice prompts. Note that there is no audio problem with  
 internal
 access to the same recording. Neither is there a problem with calls  
 not
 involving the playing of recordings. The problem occurs  
 consistently and is
 not related to system load. According to Eicon support:

 Asterisk is sending it's data packets (160 bytes each, i.e. 20ms)  
 in too
 large intervals. This causes the transmitter of the Diva Server  
 card to
 underrun and thus to fill with idle samples in regular intervals.  
 It's
 almost between any two packets where we have to insert samples.
 ...
 I wonder if anyone could provide any advice on how to continue
 troubleshooting this issue?
 I never heard of that problem before. Which versions of asterisk and
 chan_capi (I assume you use chan_capi) do you use?

 If possible, can you provide a trace with
   set verbose 9
   capi debug
 to me directly (not on the list, it is very big).
 Also, a full ditrace would help too.

 Armin
 
 
 I saw something similar with mISDN.
 If your recordings aren't an exact multiple of 20ms then
 asterisk sends a short frame for the last one.
 
 Here's a quick hack that fixed the problem for me,
 record the files in GSM. This forces the length of the recording to be
 an exact multiple of 20ms.

Hmmm, sounds the same as what I had, but it was the transcoding that
fixed it.  I actually wrote a patch to send packets through a smoother
which fixed it, and use that on all sites with IAX phones.

- --
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-users] Simultaneous Callback?!

2008-01-08 Thread Tim H. Panton

You could hack it up by dropping them both into the same conference.
You'd have to tweak the messages and other conference settings, but it would
certainly work. Not as efficient as bridging though.

Tim.

- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 08 January 2008 20:43:41 o'clock (GMT) Europe/London
Subject: [asterisk-users] Simultaneous Callback?!

We're doing callback here. Asterisk dials a number, waits for an answer, plays 
a prompt, dials a second number, and bridges the channels together.
Calls are initiated from the AMI.
No problems there. Easy stuff.

However, I'd like to know if it's possible to have Asterisk dial the same two 
numbers simultaneously, play the prompt to the first one that answers, dial the 
second one and bridge the two channels together.?

I'm not even sure how this would work within the limits of the dial plan. 
Normally, the dialing of the first leg is implicit (the channel in an AMI 
originate command), that is, there is no dial plan code for it, although you 
can specify a Local channel and asterisk will then jump into the dial plan to 
dial the first number. Once the first one answers, Asterisk jumps to the 
location specified by the second number (ie [EMAIL PROTECTED]) and calls it, 
and bridges them together.

How would this work with simultaneous numbers?
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[asterisk-users] CallerID Number incorrect in SIP packet

2008-01-08 Thread Lutgring, Sam

I am having an issue with the CallerID Number not being passed to my
phone in the SIP packet.  The CallerID Name is passed just fine and
displayed on the phone with no issue.  I have done a NoOp() in my
extension.conf and successfully seen both the CallerID name and number
correctly. So that leads me to believe that Asterisk is handeling it
correctly.  However, when I do a packet capture of the SIP packet sent
from the Asterisk server to the phone, I do not see the CallerID Number
but instead see the registered user name of the phone:
 
 
 
The lutgrins-G-2433 is the user name that my phone is registered as.  I
would expect to see sip:[EMAIL PROTECTED] instead of what I am
seeing.  Both the phone and the server are running on the same network
segment (no NAT involved).
 
Any help would be appreciated.
 
I am running Asterisk version 1.4.11
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Re: [asterisk-users] CallerID Number incorrect in SIP packet

2008-01-08 Thread Adam Moffett

in sip.conf under the definition for the sip user add
callerid=whatever

  - Original Message - 
  From: Lutgring, Sam 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, January 08, 2008 4:37 PM
  Subject: [asterisk-users] CallerID Number incorrect in SIP packet

  I am having an issue with the CallerID Number not being passed to my phone in 
the SIP packet.  The CallerID Name is passed just fine and displayed on the 
phone with no issue.  I have done a NoOp() in my extension.conf and 
successfully seen both the CallerID name and number correctly. So that leads me 
to believe that Asterisk is handeling it correctly.  However, when I do a 
packet capture of the SIP packet sent from the Asterisk server to the phone, I 
do not see the CallerID Number but instead see the registered user name of the 
phone:

  The lutgrins-G-2433 is the user name that my phone is registered as.  I would 
expect to see sip:[EMAIL PROTECTED] instead of what I am seeing.  Both the 
phone and the server are running on the same network segment (no NAT involved).

  Any help would be appreciated.

  I am running Asterisk version 1.4.11

--

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--

  No virus found in this incoming message.
  Checked by AVG Free Edition. 
  Version: 7.5.516 / Virus Database: 269.17.13/1213 - Release Date: 1/7/2008 
9:14 AM
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[asterisk-users] get_data

2008-01-08 Thread Charlie Farinella

I am calling get_data from an agi script using Asterisk::AGI like so:

$AGI-get_data('enter-conf-pin-number');

..and I am expecting to hear the file play back when I call.  I do not.  

My log entry looks like this:

-- Launched AGI Script /var/lib/asterisk/agi-bin/pbx_dev.agi
   pbx_dev.agi: CALLERID IS: XX
-- SIP/#-089e50f0 Playing 'enter-conf-pin-number' (language 'en')
-- AGI Script Executing Application: (Playback) Options: 
(is-now-being-recorded)
-- SIP/#-089e50f0 Playing 'is-now-being-recorded' (language 'en')
-- SIP/#-089e50f0 Playing 'beep' (language 'en')
[Jan  8 17:11:14] DEBUG[10646]: res_agi.c:1860 run_agi: SIP/#-089e50f0 
hungup

The log indicates that the file is played, yet that is not the case.  
Can anyone provide troubleshooting tips?

thanks,

--charlie

-- 

Charles Farinella 
Appropriate Solutions, Inc. (www.AppropriateSolutions.com)
[EMAIL PROTECTED]
voice: 603.924.6079   fax: 603.924.8668


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[asterisk-users] Linksys SPA-9xx Audio Issues

2008-01-08 Thread Andrew Joakimsen

Anyone else have problems with phones like SPA-922, SPA-921, etc?
Inbound audio is perfect but the remote end reports audio quality
issues on the audio the handset is sending out. It's not the
network I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793 
G729. Ulaw seems to be the least problematic but its still an issue.
Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
I don't know it if happens all the time but about 40% of the time the
remote caller reports they cannot hear me.

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[asterisk-users] tale of two firewalls

2008-01-08 Thread Robert Moskowitz

I have a server behind a firewall.  It is publicly addressed.  Should 
NOT be trying to NAT (how would I know).

The connection is a SIP trunk to Broadvoice.  I am calling the 
Broadvoice # from my cell and the call is being routed to my server.

With one firewall the INVITE contains information for the RTP session to 
be with broadvoice servers with different addresses.  It works.

With the other firewall the INVITE does NOT contain any other IP 
addresses, and the call goes through but no voice (duh).

I have captures of both.  I would include them in this message, but I am 
a little concerned that if some of you get the Registration, you will 
crack my secret...

PLEASE help me out on this.  I am absolutely pulling out my hair on it 
(and I don't have much).  I have stared at the Wireshark displays and 
just don't see it.  I have turned on logging on the failing firewall, 
and am not seeing any messages being dropped or rejected.



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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

2008-01-08 Thread Jared Smith

On Tue, 2008-01-08 at 17:26 -0500, Andrew Joakimsen wrote:
 Anyone else have problems with phones like SPA-922, SPA-921, etc?

If I remember correctly, the SPA-9XX phones default to sending packets
every 30ms intead of every 20ms.  Log in as Admin, click on the Advanced
link, and go to the SIP tab.  You'll find a setting labeled RTP Packet
Size.  Change it from 0.030 to 0.020 and see if that makes your
audio quality better.  It's done wonders for me in the past.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

We also use the Linksys SPA IP phones for our clients. We always change this 
setting to 0.020, which vastly improves audio performance.

What are peoples thoughts on changing it to something lower, e.g. 0.010?


Thanks,

Daniel Cole


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Wednesday, 9 January 2008 9:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues

On Tue, 2008-01-08 at 17:26 -0500, Andrew Joakimsen wrote:
 Anyone else have problems with phones like SPA-922, SPA-921, etc?

If I remember correctly, the SPA-9XX phones default to sending packets every 
30ms intead of every 20ms.  Log in as Admin, click on the Advanced link, and go 
to the SIP tab.  You'll find a setting labeled RTP Packet Size.  Change it 
from 0.030 to 0.020 and see if that makes your audio quality better.  It's 
done wonders for me in the past.

--
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

2008-01-08 Thread Andrew Joakimsen

Yep it was set to 0.030.. but the odd thing is the issue is random and
also whenever I call my mobile phones to test it seems to work fine on
the old setting.

On Jan 8, 2008 5:48 PM, Jared Smith [EMAIL PROTECTED] wrote:
 On Tue, 2008-01-08 at 17:26 -0500, Andrew Joakimsen wrote:
  Anyone else have problems with phones like SPA-922, SPA-921, etc?

 If I remember correctly, the SPA-9XX phones default to sending packets
 every 30ms intead of every 20ms.  Log in as Admin, click on the Advanced
 link, and go to the SIP tab.  You'll find a setting labeled RTP Packet
 Size.  Change it from 0.030 to 0.020 and see if that makes your
 audio quality better.  It's done wonders for me in the past.

 --
 Jared Smith
 Community Relations Manager
 Digium, Inc.


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Re: [asterisk-users] Distorted audio over Eicon Diva Server BRI

2008-01-08 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

CSB wrote:
 Sounds very similar to an issue I was having.

 Are you using mISDN?

 No. Incidentally, what's the benefit of using mISDN?

Just that its in tree and what Digium recommends for the b410p.

I'm still not 100% about it as there seems to be some introduced delay
which has meant we had to install OctasicEC for echo can as the on board
hardware one wasn't doing its job.

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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[asterisk-users] debugging bluetooth communication using chan_mobile

2008-01-08 Thread Emmanuel Favre-Nicolin

Hi,

I'm trying to setup a mobile (ericsson W300i) and I'm having some difficulties 
(to pass DTMF through the mobile and to get sound). I'd like too know how 
could debug what are the common way to debug get information.

remote mobile = mobile on asterisk (by bluetooth) = asterisk

I'd like to be able use password on asterisk. I tried and looks likes DTMF is 
not passing through the mobile on asterisk?
How am I supposed to do that?


I would also like to know the way to spy on the communication between asterisk 
and the mobile?

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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

Can you describe the issue more please? Can the remote person not hear you at 
all? Or is there distorted/broken voice?


Cheers,

Daniel Cole


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Wednesday, 9 January 2008 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Linksys SPA-9xx Audio Issues

Anyone else have problems with phones like SPA-922, SPA-921, etc?
Inbound audio is perfect but the remote end reports audio quality issues on the 
audio the handset is sending out. It's not the network I've tried asterisk 
1.2, 1.4. I've used ulaw, G726, G793  G729. Ulaw seems to be the least 
problematic but its still an issue.
Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
I don't know it if happens all the time but about 40% of the time the remote 
caller reports they cannot hear me.

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[asterisk-users] txfax_exec: Transmission loop error

2008-01-08 Thread Roger Schreiter

Hi,

I just installed Antonio Gallo's agx-ast-addons package
in order to use app_txfax with asterisk-1.4.

Compiling according to docs went well.
However, I'm getting an error after the first page
of fax:

/usr/src/agx-ast-addons/app_txfax.c:438 txfax_exec:
Transmission loop error

The (very first) page is transferred perfect anyway.
Then app_txfax unfortunetly stops the transmission.

Any hints?


Regards,
Roger.


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Re: [asterisk-users] BLF trouble

2008-01-08 Thread Lars Bensmann

On Tue, Jan 08, 2008 at 09:47:40PM +0100, Johansson Olle E wrote:
 There is a setting for enforcing the call limit for both inbound and
 outbound on a peer only.

Thanks for pointing me in the right direction. The limitonpeers=yes was
already set as I read in the documentation. But I set it in each friend
section.

Now I have moved it into the general section ...

 That way you will have BLF for both directions on a subscription.

... and it actually works after this change.

Thanks a lot,
Lars

-- 
...It's stupid to say that computer games have bad influence on childern.
If Pac-Man had influenced children born in the 80's, today we'd have lots
of kids running around in dark rooms eating pills, while listening to
monotonous and dull electronic music...
  -- Kristian Wilson, Nintendo, Inc. 1989

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[asterisk-users] Dialplan Recordings

2008-01-08 Thread Shane D

Hello,

What is the maximum WAV specs that can be used with asterisk
recordings for the Background() application?

Also, is there a place where someone can provide a custome dialplan
autoattendent for free?

-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

2008-01-08 Thread Raj Jain

This issue of phone vendors not supporting OPTIONS according to RFC 3261
often comes up on this list. Like Kevin Fleming said, an OPTIONS request is
supposed to be responded in the same way as an INVITE. Almost all SIP phone
vendors have construed OPTIONS as some kind of a keep-alive request, which
is wrong. 

Can we ask the phone vendors to play by the book?
 
--
Raj
 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Tuesday, January 08, 2008 7:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to check if a SIP phone is
forwardedwithout ringing it ?


2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]: 

Olivier wrote:

 Is there way for an Asterisk server to check if a sip
phone is forwarded
 without bothering phone's user ?

No.

 I was thinking of some Alert-Info option that would let
the phone reply 
 with a 302 Moved Temporarily or 182 Queued message and not
let the phone
 ring or display anything on its screen.

According to the SIP RFC, a SIP endpoint is supposed to
respond to an
OPTIONS message the same way that it would respond to an
INVITE message 
with the identical destination, but I've never seen a phone
respond to
an OPTIONS message with anything but '200 OK', even when a
redirect
(forward) is in place.


So, the alternative option is to play with html and use phone
embedded html server to get this redirection data. 

Cheers



--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM) 






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Re: [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...

2008-01-08 Thread Douglas Garstang

Replying to myself. :)
I just noticed the deadlock message still displayed on the console at the end 
of a normal call, so the the deadlock message is not related to the early CANCEL

- Original Message 
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 8, 2008 5:31:12 PM
Subject: [asterisk-users] Help! channel_find_deadlocked: Avoided initial 
deadlock for ...

Hope someone can help.

I have a situation where asterisk is sending a SIP CANCEL message before the 
Dial() timeout has hit. It doesn't always do it.

Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 
180 Ringing, or 183 Session Progress. It seems to be at this point that 
Asterisk starts the dial timer. Normally, when no more replies have been 
received by the dial timeout, Asterisk sends a CANCEL message. That's all fine, 
and when this happens, this is what appears on the console:

-- Called [EMAIL PROTECTED]
-- SIP/teleglobe-09879188 is making progress passing it to 
SIP/teleglobe-09876568
-- Nobody picked up in 4 ms
-- Executing
 PlayTones(SIP/teleglobe-09876568, congestion) in new stack

However, when asterisk sends the CANCEL earlier then this, this is what appears 
on the console:

-- SIP/teleglobe-09879188 is making progress passing it to 
SIP/teleglobe-09876568
  == Spawn extension (default, callback, 7) exited non-zero on 
'SIP/teleglobe-09876568'
Jan  9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided 
initial deadlock for '0x97f24d8', 10 retries!

Does anyone know what the deadlock message is all about? It is ocurring quite 
frequently.
This is Asterisk 1.2.14.

Thanks,
Doug

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[asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...

2008-01-08 Thread Douglas Garstang

Hope someone can help.

I have a situation where asterisk is sending a SIP CANCEL message before the 
Dial() timeout has hit. It doesn't always do it.

Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 
180 Ringing, or 183 Session Progress. It seems to be at this point that 
Asterisk starts the dial timer. Normally, when no more replies have been 
received by the dial timeout, Asterisk sends a CANCEL message. That's all fine, 
and when this happens, this is what appears on the console:

-- Called [EMAIL PROTECTED]
-- SIP/teleglobe-09879188 is making progress passing it to 
SIP/teleglobe-09876568
-- Nobody picked up in 4 ms
-- Executing PlayTones(SIP/teleglobe-09876568, congestion) in new stack

However, when asterisk sends the CANCEL earlier then this, this is what appears 
on the console:

-- SIP/teleglobe-09879188 is making progress passing it to 
SIP/teleglobe-09876568
  == Spawn extension (default, callback, 7) exited non-zero on 
'SIP/teleglobe-09876568'
Jan  9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided 
initial deadlock for '0x97f24d8', 10 retries!

Does anyone know what the deadlock message is all about? It is ocurring quite 
frequently.
This is Asterisk 1.2.14.

Thanks,
Doug







  

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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

2008-01-08 Thread Kev S

The issues i have been having are probably similar to the original 
message, I use the Linksys 9XX Series phones and we used to always 
receive complaints from the person we were calling that they could 
hardly hear us.

I fixed this by:

Going into the Phone section of the config and setting the Handset, 
Speakerphone and Headset input gain to 6.

And i also went into SIP and changed the RTP Packet Size to 0.020

This resolved the low volume issue, Sorry if you have a no sound issue, 
but thats how i resolved very low volume.

Phones sound great now!

Regards,
Kevin Sandalin

Daniel Cole wrote:
 Can you describe the issue more please? Can the remote person not hear you at 
 all? Or is there distorted/broken voice?


 Cheers,

 Daniel Cole


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew 
 Joakimsen
 Sent: Wednesday, 9 January 2008 9:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Linksys SPA-9xx Audio Issues

 Anyone else have problems with phones like SPA-922, SPA-921, etc?
 Inbound audio is perfect but the remote end reports audio quality issues on 
 the audio the handset is sending out. It's not the network I've tried 
 asterisk 1.2, 1.4. I've used ulaw, G726, G793  G729. Ulaw seems to be the 
 least problematic but its still an issue.
 Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
 I don't know it if happens all the time but about 40% of the time the remote 
 caller reports they cannot hear me.

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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

I have found with a number of clients to who we have installed the LinkSys 
phones, that when you get the input gains to 6, that the phones have a tendency 
to pick up too much background noise. Have you experienced this at all?

Cheers,

Daniel Cole


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S
Sent: Wednesday, 9 January 2008 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues

The issues i have been having are probably similar to the original message, I 
use the Linksys 9XX Series phones and we used to always receive complaints from 
the person we were calling that they could hardly hear us.

I fixed this by:

Going into the Phone section of the config and setting the Handset, 
Speakerphone and Headset input gain to 6.

And i also went into SIP and changed the RTP Packet Size to 0.020

This resolved the low volume issue, Sorry if you have a no sound issue, but 
thats how i resolved very low volume.

Phones sound great now!

Regards,
Kevin Sandalin

Daniel Cole wrote:
 Can you describe the issue more please? Can the remote person not hear you at 
 all? Or is there distorted/broken voice?


 Cheers,

 Daniel Cole


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
 Joakimsen
 Sent: Wednesday, 9 January 2008 9:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Linksys SPA-9xx Audio Issues

 Anyone else have problems with phones like SPA-922, SPA-921, etc?
 Inbound audio is perfect but the remote end reports audio quality issues on 
 the audio the handset is sending out. It's not the network I've tried 
 asterisk 1.2, 1.4. I've used ulaw, G726, G793  G729. Ulaw seems to be the 
 least problematic but its still an issue.
 Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
 I don't know it if happens all the time but about 40% of the time the remote 
 caller reports they cannot hear me.

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Re: [asterisk-users] tale of two firewalls





robert,
with limited info below, are you port forwarding on the router with the
public IP, ports 10,000-20,000, 5004, along with 5060? and the other
router (internal, I assume)???

how do you have two firewalls configured with one * box?
do you have captures on both sides of the internal (I assume)
router?... 
if you want to call me, just send me an email... will be available till
11p EST.
daveC

Robert Moskowitz wrote:

  I have a server behind a firewall.  It is publicly addressed.  Should 
NOT be trying to NAT (how would I know).

The connection is a SIP trunk to Broadvoice.  I am calling the 
Broadvoice # from my cell and the call is being routed to my server.

With one firewall the INVITE contains information for the RTP session to 
be with broadvoice servers with different addresses.  It works.

With the other firewall the INVITE does NOT contain any other IP 
addresses, and the call goes through but no voice (duh).

I have captures of both.  I would include them in this message, but I am 
a little concerned that if some of you get the Registration, you will 
crack my secret...

PLEASE help me out on this.  I am absolutely pulling out my hair on it 
(and I don't have much).  I have stared at the Wireshark displays and 
just don't see it.  I have turned on logging on the failing firewall, 
and am not seeing any messages being dropped or rejected.



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-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894






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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

2008-01-08 Thread Kev S

No, I haven't experienced this.

I think were lucky because most voip phones are in there own offices, I 
will check with our sales manager this afternoon who sits in the call 
center and see what the background noise is like on her phone.

I guess i'm just lucky that its a quiet environment, But there are a few 
people who *may* be affected and i will check this out and let you know.

Regards,
Kevin

Daniel Cole wrote:
 I have found with a number of clients to who we have installed the LinkSys 
 phones, that when you get the input gains to 6, that the phones have a 
 tendency to pick up too much background noise. Have you experienced this at 
 all?

 Cheers,

 Daniel Cole


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S
 Sent: Wednesday, 9 January 2008 12:35 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues

 The issues i have been having are probably similar to the original message, I 
 use the Linksys 9XX Series phones and we used to always receive complaints 
 from the person we were calling that they could hardly hear us.

 I fixed this by:

 Going into the Phone section of the config and setting the Handset, 
 Speakerphone and Headset input gain to 6.

 And i also went into SIP and changed the RTP Packet Size to 0.020

 This resolved the low volume issue, Sorry if you have a no sound issue, but 
 thats how i resolved very low volume.

 Phones sound great now!

 Regards,
 Kevin Sandalin

 Daniel Cole wrote:
   
 Can you describe the issue more please? Can the remote person not hear you 
 at all? Or is there distorted/broken voice?


 Cheers,

 Daniel Cole


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
 Joakimsen
 Sent: Wednesday, 9 January 2008 9:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Linksys SPA-9xx Audio Issues

 Anyone else have problems with phones like SPA-922, SPA-921, etc?
 Inbound audio is perfect but the remote end reports audio quality issues on 
 the audio the handset is sending out. It's not the network I've tried 
 asterisk 1.2, 1.4. I've used ulaw, G726, G793  G729. Ulaw seems to be the 
 least problematic but its still an issue.
 Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
 I don't know it if happens all the time but about 40% of the time the remote 
 caller reports they cannot hear me.

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Re: [asterisk-users] :POSSIBLE SPAM: Re: conferencing help





steve,
thanks for pointing that out, I forgot the exact reason. 
as for the hearing/audio problem... if all else works the conferencing
should also... I haven't used freepbx, do they handle the port
filtering?

 # tcpdump -i eth0 udp 

should show if the packets are getting in/out...

I have no experience with sangoma cards.
daveC

Steve Edwards wrote:

  
dave cantera wrote:


  nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
using meetme conferencing.
  

  
  
On Wed, 9 Jan 2008, Nhadie wrote:

  
  
hi dave thank you for the reply. i have loaded zap and using only
ztdummy but still can't hear anything when i dial ti my conference, i
think this explains it already. will a sangoma card do?

  
  
I use ztdummy with meetme conferencing and it works fine on CentOS 4.5. 
Ztdummy is not an issue until you get xx callers in xx conferences.

I think (but have no empirical data to back it up) that a card yields 
better sound quality at higher call levels.

Thanks in advance,

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

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-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894






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Re: [asterisk-users] Dialplan Recordings

2008-01-08 Thread Tilghman Lesher

On Tuesday 08 January 2008 19:46:50 Shane D wrote:
 What is the maximum WAV specs that can be used with asterisk
 recordings for the Background() application?

All recordings must currently be in single channel, 8kHz format.  The
maximum length of an uncompressed wav file is approximately 38
hours (due to the 32-bit headers).

-- 
Tilghman

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Re: [asterisk-users] conferencing help

2008-01-08 Thread Nhadie

Hi Steve,

I see. I have this now,

*CLI zap show channels
Chan Extension  Context Language   MusicOnHold
pseudodefault en

*CLI load chan_zap.so
Unable to load module chan_zap.so   -- on the log file it says, it as 
already loaded that's why it's unable to load.

i tried my calling to my conf 6000

 -- Executing Set(SIP/100-081825b0, MEETME_OPTS=iM) in new stack
 -- Executing Goto(SIP/100-081825b0, STARTMEETME|1) in new stack
 -- Goto (from-internal,STARTMEETME,1)
 -- Executing MeetMe(SIP/100-081825b0, 6000|iM|) in new stack
   == Parsing '/etc/asterisk/meetme.conf': Found
   == Parsing '/etc/asterisk/meetme_additional.conf': Found
 -- Created MeetMe conference 1023 for conference '6000'
 -- Recording
 -- Playing 'vm-rec-name' (language 'en')
 -- Executing Set(SIP/100-081825b0, MEETME_OPTS=iM) in new stack
 -- Executing Goto(SIP/100-081825b0, STARTMEETME|1) in new stack
 -- Goto (from-internal,STARTMEETME,1)
 -- Executing MeetMe(SIP/100-081825b0, 6000|iM|) in new stack
   == Parsing '/etc/asterisk/meetme.conf': Found
   == Parsing '/etc/asterisk/meetme_additional.conf': Found
 -- Created MeetMe conference 1023 for conference '6000'
 -- Recording
 -- Playing 'vm-rec-name' (language 'en')


it's trying to play something 'vm-rec-name' but i cannot hear anything 
on the phone. i'm using g711. i'm not using trixbox, i just installed 
asterisk, freepbx, zaptel, etc on a debian box. i'm using all the latest 
version i downloaded from the website (i used asterisk 1.2).

/usr/include# modprobe -l | grep ztdum
/lib/modules/2.6.18-5-686/misc/ztdummy.ko

/usr/include# modprobe -l | grep zap
/lib/modules/2.6.18-5-686/misc/zaptel.ko

how do i know if my ztdummy is working properly? thanks again!

regards,
nhadie





Steve Edwards wrote:
 dave cantera wrote:
 nhadie,
 meetme requires a zaptel timing device... ztdummy is unreliable when
 using meetme conferencing.
 
 On Wed, 9 Jan 2008, Nhadie wrote:
 
 hi dave thank you for the reply. i have loaded zap and using only
 ztdummy but still can't hear anything when i dial ti my conference, i
 think this explains it already. will a sangoma card do?
 
 I use ztdummy with meetme conferencing and it works fine on CentOS 4.5. 
 Ztdummy is not an issue until you get xx callers in xx conferences.
 
 I think (but have no empirical data to back it up) that a card yields 
 better sound quality at higher call levels.
 
 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] conferencing help

2008-01-08 Thread Matt Riddell

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Nhadie wrote:
 Hi Steve,
 
 I see. I have this now,
 
 *CLI zap show channels
 Chan Extension  Context Language   MusicOnHold
 pseudodefault en

That means the zap channel should be ok.

One thing you could do is go to the place you downloaded Zaptel and type:

./zttest -v

Do you get numbers (i.e. something close or closish to 100%)?

Also, if you just have the extensions:

exten = 555,1,Answer()
exten = 555,n,Background(demo-echotest)
exten = 555,n,Echo()

Do you get an answer?

You don't really need the brackets on answer and echo but I usually type
that way and then add options.  :-)


- --
Kind Regards,

Matt Riddell
Director
___

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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFHhFUKDQNt8rg0Kp4RAtfQAKC/mjeswAVxnkzv/HHC/4ZCL92SEwCfRoY7
rIAGfpE/0dh56i9myEbOFfA=
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Re: [asterisk-users] conferencing help

2008-01-08 Thread gary

I will be out of the office on Wednesday, January 9, 2008.  If this is an 
emergency, please call Customer Service at (877) 791-7700.  Thank you.


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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

Ok, no worries :)

Most of our clients have a relatively open common work area, where the phones 
are located. I would be interested to know what your sales manager has 
experienced.


Cheers,

Daniel Cole


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S
Sent: Wednesday, 9 January 2008 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues

No, I haven't experienced this.

I think were lucky because most voip phones are in there own offices, I will 
check with our sales manager this afternoon who sits in the call center and see 
what the background noise is like on her phone.

I guess i'm just lucky that its a quiet environment, But there are a few people 
who *may* be affected and i will check this out and let you know.

Regards,
Kevin

Daniel Cole wrote:
 I have found with a number of clients to who we have installed the LinkSys 
 phones, that when you get the input gains to 6, that the phones have a 
 tendency to pick up too much background noise. Have you experienced this at 
 all?

 Cheers,

 Daniel Cole


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Kev S
 Sent: Wednesday, 9 January 2008 12:35 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Linksys SPA-9xx Audio Issues

 The issues i have been having are probably similar to the original message, I 
 use the Linksys 9XX Series phones and we used to always receive complaints 
 from the person we were calling that they could hardly hear us.

 I fixed this by:

 Going into the Phone section of the config and setting the Handset, 
 Speakerphone and Headset input gain to 6.

 And i also went into SIP and changed the RTP Packet Size to 0.020

 This resolved the low volume issue, Sorry if you have a no sound issue, but 
 thats how i resolved very low volume.

 Phones sound great now!

 Regards,
 Kevin Sandalin

 Daniel Cole wrote:

 Can you describe the issue more please? Can the remote person not hear you 
 at all? Or is there distorted/broken voice?


 Cheers,

 Daniel Cole


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Andrew
 Joakimsen
 Sent: Wednesday, 9 January 2008 9:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Linksys SPA-9xx Audio Issues

 Anyone else have problems with phones like SPA-922, SPA-921, etc?
 Inbound audio is perfect but the remote end reports audio quality issues on 
 the audio the handset is sending out. It's not the network I've tried 
 asterisk 1.2, 1.4. I've used ulaw, G726, G793  G729. Ulaw seems to be the 
 least problematic but its still an issue.
 Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
 I don't know it if happens all the time but about 40% of the time the remote 
 caller reports they cannot hear me.

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[asterisk-users] Register source port

2008-01-08 Thread Al lists

Hello all,
is there any way to tell asterisk what port to use for source of any
registration request?
for example the simple register command,
register = user:[EMAIL PROTECTED]:port
will send the register packet from asterisk_IP:5060 to proxy:port .
Is there anyway to have asterisk to use different port instead of 5060 for
each register command, like 5060 for the first 5070 for second .. ?
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Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

As using OPTIONS requests main benefit is to non-phone specific, what shall
we do when most vendors do not comply with RFC ?

2008/1/9, Raj Jain [EMAIL PROTECTED]:

 This issue of phone vendors not supporting OPTIONS according to RFC 3261
 often comes up on this list. Like Kevin Fleming said, an OPTIONS request
 is
 supposed to be responded in the same way as an INVITE. Almost all SIP
 phone
 vendors have construed OPTIONS as some kind of a keep-alive request, which
 is wrong.

 Can we ask the phone vendors to play by the book?

 --
 Raj


 

 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Olivier
 Sent: Tuesday, January 08, 2008 7:50 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] How to check if a SIP phone is
 forwardedwithout ringing it ?


 2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]:

 Olivier wrote:

  Is there way for an Asterisk server to check if a sip
 phone is forwarded
  without bothering phone's user ?

 No.

  I was thinking of some Alert-Info option that would let
 the phone reply
  with a 302 Moved Temporarily or 182 Queued message and
 not
 let the phone
  ring or display anything on its screen.

 According to the SIP RFC, a SIP endpoint is supposed to
 respond to an
 OPTIONS message the same way that it would respond to an
 INVITE message
 with the identical destination, but I've never seen a
 phone
 respond to
 an OPTIONS message with anything but '200 OK', even when a
 redirect
 (forward) is in place.


 So, the alternative option is to play with html and use phone
 embedded html server to get this redirection data.

 Cheers



 --
 Kevin P. Fleming
 Director of Software Technologies
 Digium, Inc. - The Genuine Asterisk Experience (TM)






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Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

2008/1/8, Jean-Louis curty [EMAIL PROTECTED]:

 Hi,

 I succesfully install spandsp chan_misdn and digium card. the rxfax works
 fine and I get the fax result by email.
 I would like to do the same using a Patton gw + zaptel but I can't receive
 fax anymore,


which patton product do you use ?
how are patton gw and asterisk connected to each other ?

the call comes in from ISDN in the Patton gw, patton sends it to asterisk,
 asterisk run a macro to make a tif file using rxfax,
 the tif file is correctly created but with a 0 size the call looks normal,
 1 pages, 45 seconds and disconnect but the file is still 0,

 anyone succeeded in this ?
 Many (many) thanks!
 jean-louis

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Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?


9 jan 2008 kl. 02.48 skrev Raj Jain:

 This issue of phone vendors not supporting OPTIONS according to RFC  
 3261
 often comes up on this list. Like Kevin Fleming said, an OPTIONS  
 request is
 supposed to be responded in the same way as an INVITE. Almost all  
 SIP phone
 vendors have construed OPTIONS as some kind of a keep-alive request,  
 which
 is wrong.
Which we do too, by the way. In worst case, maybe Asterisk has set  
this industry
standard.

OPTIONS is far to heavy in processing on the server side to be used  
for keep-alives. I'm  starting to see devices that use it for checking  
capabilities - the proper way. To do this properly, we will have to  
authenticate the OPTIONs request and match it with the proper peer/ 
user to get the proper codec settings, ACLs and such.

Since all versions of Asterisk use OPTIONs for NAT-keepalives, I'm a  
bit hesitant to fix this. It's a catch 22. I want to do it properly,  
but then the amount of processing for each OPTIONs request that we  
receive is going to be a bit too much. Maybe one could ask vendors to  
add a header to the  OPTIONs packet saying this is just a keep-alive.  
Give me a 200 OK without any parsing and be happy, because I don't  
care about the reply.

Linksys has a setting and use NOTIFY for Keep-alives, which also is a  
poor solution, but at least something we can just give an error  
response to without a lot of processing. There was a proposal for  
PING, but it never got anywhere.

/O

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Re: [asterisk-users] Register source port