re: [asterisk-users] error when compiling zaptel-1.4

[Alyed Tzompa]

The error lies here:

make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'

make: *** arch/i386/boot: No such file or directory.  Stop.

do you have the kernel-headers installed? (e.g.  
glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora) 

Alyed 


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when i compile zaptel
make linux26
make install
i got these errors:

make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory.  Stop.
make: Entering an unknown directorymake: Leaving an unknown
directorymake[2]: *** [archclean] Error 2
make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386'
make[1]: *** [clean] Error 2
make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods'
make: *** [clean] Error 2

any idea
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re: [asterisk-users] Automatic Dial, Play message

[Alyed Tzompa]

I've made a very simple one time ago I
could share with you, it's made on bash, takes as input a CSV file,
places the calls using the /var/spool/asterisk/outbound directory, and
restricts the number of calls to a given number at a time (say 10)

I can share it with you only if you promise NOT to use it for
Telemarketing, otherwise might the mighty spirits of (place the name of
some super natural power here) cast the most terrible spell on you
forever and ever.

so, if you still want it just contact me directly :)

Alyed  


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Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?

Just for example, you have a database of

FirstName, LastName, PhoneNumber
Jon, Beck, 9194713175

So it would pull each record with phone number, dial the number, when
answered play a pre-recorded message.

It could be used to notify parents at a school that a after school
game is canceled.

I appreciate any direction you can point me in.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]

-- 
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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re: [asterisk-users] Asterisk outbound calling does not wait for answer before playback

[Alyed Tzompa]

Had the same issue time ago, but Eric shed good light on it, 
have a look at:

http://lists.digium.com/pipermail/asterisk-users/2006-November/172079.html

Summary: sorry, no nice work around.

Alyed  


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Hello Asteriskers, :-)

We're trying to set up an outbound notification calling for system 
alerts with Asterisk 1.4.0.  We generate a call file in 
/var/spool/asterisk/outgoing and the outbound call is originated through 
Zap/1 (Sangoma A200D to a Canadian POTS line).  The problem is that 
Asterisk does not wait for the other side to answer before it starts 
playing the message.  So the person called answers the phone after the 
second or third ring and only hears the tail end of the message and the 
goodbye.

Ideally, we want to deliver the message immediately after the person 
answers, or if an answering machine picks up, right after the beep.

Any suggestions?

(1) The call file generator script (works ok):
#!/bin/sh

TMPFILE=`mktemp /tmp/tmp.XXX` || exit 1
echo TMPFILE = $TMPFILE

cat  $TMPFILE
Channel: Zap/g1/phone_number_here
Callerid: SYSTEM
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: dialout
Extension: s
Priority: 1
EOT

mv -v $TMPFILE /var/spool/asterisk/outgoing

(2) The dialout context in extensions.conf (problem - starts playback 
before call is answered)
[dialout]
exten = s,1,NoOp(Dialout)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=8)
exten = s,n,Set(MACHINE=0)
exten = s,n,Answer
exten = s,n,BackgroundDetect(silence/5,1000,50)
exten = s,n,NoOp(Ans Machine detected)
exten = s,n,Set(MACHINE=1)
exten = s,n,BackgroundDetect(silence/30,1000,50,30050)
exten = s,n,NoOp(Ans Machine Message Too Long)
exten = s,n,Hangup

exten = talk,1,GotoIf($[${MACHINE}=1]?machine:human)
exten = talk,2(machine),Goto(dialout-machine,s,1)
exten = talk,3(human),Goto(dialout-human,s,1)

[dialout-machine]
exten = s,1,NoOp(Dialout to Ans Machine)
exten = s,n,Playback(/tmp/asterisk-recording)
exten = s,n,Wait(1)
; we'd like to do something to wait for the beep here...
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup

[dialout-human]
exten = s,1,NoOp(Dialout to Human)
exten = s,n,Playback(/tmp/asterisk-recording)
exten = s,n,Wait(1)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup

(3) *CLI
-- Attempting call on Zap/1/1234567 for [EMAIL PROTECTED]:1 (Retry 1)
Channel Zap/1-1 was answered.
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, Dialout) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(Zap/1-1, TIMEOUT(digit)=5) in new 
stack
-- Digit timeout set to 5
-- Executing [EMAIL PROTECTED]:3] Set(Zap/1-1, TIMEOUT(response)=8) in 
new stack
-- Response timeout set to 8
-- Executing [EMAIL PROTECTED]:4] Set(Zap/1-1, MACHINE=0) in new stack
-- Executing [EMAIL PROTECTED]:5] Answer(Zap/1-1, ) in new stack
(Problem: Asterisk does not wait until the call is answered on the far end!)
-- Executing [EMAIL PROTECTED]:6] BackgroundDetect(Zap/1-1, 
silence/5|1000|50) in new stack
-- Playing 'silence/5' (language 'en')
-- Executing [EMAIL PROTECTED]:1] GotoIf(Zap/1-1, 0?machine:human) 
in new stack
-- Goto (dialout,talk,3)
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, dialout-human|s|1) 
in new stack
-- Goto (dialout-human,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, Dialout to Human) 
in new stack
-- Executing [EMAIL PROTECTED]:2] 
Playback(Zap/1-1,/tmp/asterisk-recording) in new stack
-- Playing '/tmp/asterisk-recording' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:4] Playback(Zap/1-1, vm-goodbye) 
in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing [EMAIL PROTECTED]:5] Hangup(Zap/1-1, ) in new stack
  == Spawn extension (dialout-human, s, 5) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
[Feb  8 13:29:37] NOTICE[32512]: pbx_spool.c:351 attempt_thread: Call 
completed to Zap/1/1234567

Thanks for any ideas on this!

Alvin

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Re: [asterisk-users] Sangoma A101 with Unicall

[Alyed Tzompa]

Carlos: Had you tried re-compiling wanpipe?

Had a similar problem, and eventhough I'm pretty sure I compiled
wanpipe and it did the zaptel patch succesfully, once I finished with
the Unicall installation, somehow the patch was not wotking correctly.
So after a couple of days of looking around I decided to re-compile
wanpipe (run  ./Setup install where you untared the
wanpipe-X.X.tar.gz) and afterwards it worked fine.

if it doesn't try contacting me directly, maybe I can have an eye on your 
system and help you out.

Alyed  


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On Thu, 21 Dec 2006 08:33:27 -0500, Doug Lytle wrote
 Carlos Chavez wrote:
  CAS signalling on span 1 conflicts with HDLC with FCS check on channel
 
 
 My guess is not to use HDLC, as the error says above, that it 
 conflicts with CAS.
 

 I wish it were that easy and obvious.  I only found one setting on the
configuration to disable HDLC and it is disabled.

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[asterisk-users] mfcr/R2

[Alyed Tzompa]

Hello!

I'm tryuing to bring up an R2 connection but eventhough I've followed
the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems
to be missing.

When an incomming call is generated I get:

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24  - 0001 
[1/  
1/Idle 
/Idle]

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Detected

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Making a new call with CRN 32771

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 1101  - 
[2/  
2/Idle 
/Idle]

Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:2672 handle_uc_event: 
Unicall/24 event Detected

and that's it, afterwards just a busy tone and the telco guy says the channel 
turns sealed.

When I try an outbound call I get:

-- Attempting call on Unicall/g2/12345678 for [EMAIL PROTECTED]:1 (Retry 1)

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Call control(1)

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Make call

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Making a new call with CRN 32769

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 0001  - 
[1/  
1/Idle 
/Idle]

Nov 24 05:53:52 WARNING[-286483536]: chan_unicall.c:2672 handle_uc_event: 
Unicall/24 event Dialing

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 seize_ack_wait_expired

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 R2 prot. err. [1/ 
40/Seize
/Idle] cause 32776 - Seize ack timed out

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report:
MFC/R2 UniCall/24 1001  - 
[1/  
1/Idle 
/Idle]

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:2672 handle_uc_event: 
Unicall/24 event Protocol failure

-- Unicall/24 protocol error. Cause 32776

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Channel echo cancel

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Channel gains

Nov 24 05:53:54 WARNING[-286483536]: chan_unicall.c:612 unicall_report: MFC/R2 
UniCall/24 Channel switching

-- Hungup 'UniCall/24-1'

Nov 24 05:53:54 NOTICE[-286483536]: pbx_spool.c:234 attempt_thread: Call failed 
to go through, reason 1

Something else funny is happening: as soon as the telco makes a reset
of the trunk they say they start receving data as if the PBX would be
generating calls, then all channels go sealed except for 1.

Anyone having an idea on how to solve this?

Alyed


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re: [asterisk-users] Asterisk 1.2.13 can't load module app_curl.so

[Alyed Tzompa]

First look if you have the libidn.so.11 library. if you don't 
then install it, otherwise you can simply copy-paste it into the /usr/lib 
folder where Asterisk is looking for it or make a symbolic link to it.

Alyed 



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When i start asterisk : 
asterisk -vvvc

[app_curl.so]Nov 16 15:23:57 
WARNING[4469]: loader.c:325 __load_resource: /usr/lib/libidn.so.11: undefined 
symbol: stringprep_rfc3454_A_1
Nov 16 15:23:57 WARNING[4469]: loader.c:554 load_modules: Loading module 
app_curl.so failed!

Do you have any 
insght ?

My setup :

Fedora Core 6
Asterisk 1.2.13


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[asterisk-users] Harris picking up before extension

[Alyed Tzompa]

		Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is when dialing to the Harris PBX, it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folder I dial an extension,
say 100, and play a prompt as soon as it is picked up, the promt is
beign played as soon as it reaches the Harris, eventhough the given
extension can still be ringing. If I let the extension ring for a while
and then pick up I only hear the prompt in the middle (or as far as it
went
till I picked up).
Have tried kewlstart, loopstart, groundstart and even the
answeronpolarityswitch configs in zapata.conf but can't find the
solution.
Any one having solved this problem?Alyed
		

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[asterisk-users] Harris 20-20

[Alyed Tzompa]

		Hi there!
I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris
20-20 PBX. More less everything went fine, but the problem I have now
is that when dialing to the Harris PBX it seems to pick up my call as
soon as it reaches it.
For example if from the Asterisk outgoing folder I dial an extension,
say 100, and play it a prompt as soon as it is picked up, the promt is
beign played as soon as it reaches the Harris, eventhough the given
extension can still be ringing. If I let it ring and then pick up this
extension I only hear the prompt in the middle (or as far as it went
till I picked up).
Have tried kewlstart, loopstart, groundstart and even the
answeronpolarityswitch configs in zapata.conf but can't find the
solution.
Any one having solved this problem?Alyed
		

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

[Alyed Tzompa]

		If your Asterisk is behind a NAT you can use externip=x.x.x.x (sip.conf)
If your Sipura is behind a NAT you can use nat= yes (sip.conf)
		Btu I'm really afraid that unless you use a SIP
proxy (e.g. Portaone) you won't be able to succesfully connect both
elements if they both are behind NATs. It is just because of how SIP and NAT work together.
		
		
Alyed 
		
		
		
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		Does anybody have a good link how to connect Sipura with Asteriks, bothbehind NAT?  I'm using FWD but their connection is like a weather(especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good linkexplaining how to setup Linux server-- #Joseph___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Live creation of trunk groups

[Alyed Tzompa]

		My advice is to first make some tests to see if a reload is enough for Asterisk to read any group definitions change in zapata.conf, otherwise no on-the-fly change will workAlyed 
		
		

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		Hi,Is there a way to create trunk groups while asterisk is running.For exemple let's say that zapata.conf defines g0 as channels 1-23I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23Any hints appreciated.Andre Courchesne___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Good phones for outside of the office?

[Alyed Tzompa]

		Isn't your problem more about NAT traversal rather than the phones themselves?if so better use some iax softphone, have a look at: http://www.voip-info.org/wiki-VOIP+Phonesof course you can use SIP based hard/soft phones but using iax based ones is cheaper and faster.Alyed
		
		

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		I am looking for phones that work well (or at all) when outside of the network and behind a router, such as at someone's home or in a hotel. My Polycom IP601s do not seem to be up to the task, so I am hoping that there is a good alternative for my outside sales people to use to talk to my asterisk server.Thanks in advance,Warren___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Re: ECHO Cancellation in SIP Calls

[Alyed Tzompa]

		Echo is generated by the analog end to where you place the call, not the IP side of it.
As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it)
I'm afraid there is little you can do to here.Alyed 
		
		
		
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		On 2006-10-26 03:18:20 -0700, Stefan Agethen  said: Hi,  i am from Germany, so excuse my School English.  I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update  of Asterisk 2 wooks ago, Echos accure in my SIP Calls.  I use SNOM 360, sometimes there is no echo (for example if i call  myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing  is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS  Can i control the cancellation with the zapata.conf ?  I have tried this with "echocancel=..." and so on, no luck :(  I would be glad to get some help, the Docs of Asterisk dont explain how  to cancel Echos in ! SIP !Are you hearing the echo, or is the far end party?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Re: duplicate ghost calls with long duration

[Alyed Tzompa]

		you can also try using
busydetect=yes
busycount=4
in your zapata.conf
Hopefuly you won't start getting sudden hang ups, due to false positives and it will be helpful enough.
Alyed 
		
		
		
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		On 2006-10-17 09:00:51 -0700, Bjoern Metzdorf  said: I run into that from time to time for this business account we have  where channels were staying open for a long time so I made a script run  from cron to hang up any extension over X amount of time:  /usr/sbin/asterisk -rx "show channels concise" |awk -F : '($11  5400)  {print "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""} '|sh  This looks at any calls over 90 minutes then hangs it up. You can  modify it for your issue say something like:  /usr/sbin/asterisk -rx "show channels concise" |awk -F :  '/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11  5400) {print  "/usr/sbin/asterisk -rx \"soft hangup " $1 "\""} '|sh  Not practical though for saving money... If someone is on for say 1  minute and there is an issue with the channel not hanging up, 5399  minutes would still be billed.  What version are you using?  I never had these issues with asterisk 1.0.x in 15 months. That leads  me to a problematic 1.2.x or to faulty bristuff-patches.  I will upgrade asterisk asap to latest 1.2.x and add an absolute  timeout to those destinations.  But: Are we the only ones experiencing this?That really doesn't sound at all the same to me as what he is describing?  In his case (ie not hung up calls) if you are using SIP handsets, then the rtptimeout setting can cut the calls off when there is no audio data flowing.Good Luck,Marty___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		Hi!
You were right! changing the Calling Search Space for the SIP trunk worked!.
Many thanks for your help Eric. Now I'm fine tuning the application, but the communication with Cisco works pretty fione so far.Alyed 
		
		
		
Return-Path: [EMAIL PROTECTED] Wed Oct 11 10:57:00 2006Received: from rwcrmhc15.comcast.net [204.127.192.85] by maila11.webcontrolcenter.com with SMTP;   Wed, 11 Oct 2006 10:57:00 -0700
		
		
		I
think SIP is working on your CM becuase it sends a 404 message, I think
your problem is the CM cant find the endpoint in its database.I
am running CM 4.1(3)sr1, and under where you enter the device name
there is a section called, "Call Routing Information". There you should
see a field called "Calling Search Space".You should select the
Calling Search space that the destination endpoints are in. Or if you
want to create a new calling search space, and then give them access to
each other you can do that.In the attachment, I circled the calling search space field I see on my Add NEW SIP TRUNK PAGE.Hope this helps. -- Original message ------From: "Alyed Tzompa"   		Many thanks for your interest :)  In the Cisco the only thing I've donde so far is enable a SIP trunk and pointing  it towards the IP the Asterisk has.  For this I mean:  Device -- Trunk -- Add a new Trunk  Seleccted for Trunk Type "SIP trunk"  Selected for Device Protocol "SIP"  Entered for   		Device Name "asterisk"  		Entered for   		Description "asterisk"  Selected the appropriate device for Device Pool  Entered for Destination Address "192.168.1.20" (Asterisk's IP)  Selected for Outgoing Transport Type "UDP"  Entered the appropriate routing patterns  clicked on "Insert"  in sip.conf I'm using the same config shown in th VoIP-info wiki:  [cisco]  type=friend  context=cisco-test  host=192.168.1.100  disallow=all  allow=ulaw  allow=alaw  nat=no  canreinvite=yes  qualify=yes   Regards,  Alyed Return-Path:  Tue Oct 10 16:24:58 2006 Received: from rwcrmhc13.comcast.net [216.148.227.153] by  maila11.webcontrolcenter.com with SMTP;Tue, 10 Oct 2006 16:24:58 -0700  I can check for you tomorrow and send you screen shots.  How do you have the asterisk configured on the CM. Please send me some  configuration information.  -Eric  -- Original message -- From: "Alyed Tzompa" 		Do you know where in the Cisco menu can I  check out this "calling search spaces" you point out? I don't have  access to it right now so want to know if this is a configurable menu  in the web interface or will need to ask for expert (and expensive)  Cisco assistance.Alyed    Return-Path:  Tue Oct 10 11:04:09 2006  Received: from rwcrmhc11.comcast.net [216.148.227.151] by   maila11.webcontrolcenter.com with SMTP; Tue, 10 Oct 2006 11:04:09 -0700Looks  like the CallManager is unable to find the endpoint in its database.  Make sure asterisk trunk on the Call manager is in the same "calling  Search Space" as the phones are in, or make sure there is access  between the "calling search spaces"-Eric -- Original message --  From: "Alyed Tzompa"   		Hi!  I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've  followedthe info in  http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integrat   ion   but still not able to make Asterisk communicate with Cisco. I keep on   receiving--- 		SIP/2.0 400 Bad Request - 'Malformed/Missing URL'		--- and --- SIP/2.0 404 Not Found  --- 		messages everytime I send a call. Had play a lot with the waySIP messages are sent to the Cisco, but always been unseccessful.   I'm begining to think this is more of a Cisco config problem than   Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so   dun't know if I need to "enable" SIP messageing/reception in the Cisco.  Regards,  Alyed
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[asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		Hi!
I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in 
http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration 
but still not able to make Asterisk communicate with Cisco. I keep on receiving --- 
		SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
		--- and ---  
		
SIP/2.0 404 Not Found --- 
		messages everytime I send a call. Had play a lot with the way SIP messages are sent to the Cisco, but always been unseccessful. 
I'm begining to think this is more of a Cisco config problem than
Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so
dun't know if I need to "enable" SIP messageing/reception in the Cisco.
Regards,
		
		
		
Alyed 
		

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Re: [asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		Do you know where in the Cisco menu can I
check out this "calling search spaces" you point out? I don't have
access to it right now so want to know if this is a configurable menu
in the web interface or will need to ask for expert (and expensive)
Cisco assistance.Alyed 
		
		
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		Looks
like the CallManager is unable to find the endpoint in its database.
Make sure asterisk trunk on the Call manager is in the same "calling
Search Space" as the phones are in, or make sure there is access
between the "calling search spaces"-Eric -- Original message ------From: "Alyed Tzompa"   		Hi!  I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed  the info in   http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integrat ion   but still not able to make Asterisk communicate with Cisco. I keep on receiving  ---   		SIP/2.0 400 Bad Request - 'Malformed/Missing URL'  		--- and --- SIP/2.0 404 Not Found  ---   		messages everytime I send a call. Had play a lot with the way  SIP messages are sent to the Cisco, but always been unseccessful.   I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to "enable" SIP messageing/reception in the Cisco.  Regards,  Alyed 
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Re: [asterisk-users] Cisco CCM - Asterisk

[Alyed Tzompa]

		What I want is to transfer some calls to a Cisco extension, so think I don't need to do the upgrade to CM5.I'm I right?AlyedOn Tue, Oct 10, 2006 at 01:21:28PM -0500, Lacy Moore - Aspendora wrote: I'm begining to think this is more of a Cisco config problem than Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so dun't know if I need to "enable" SIP messageing/reception in the Cisco. Yeah, pretty sure to enable it you're going to have to upgrade to CCM 5.  I could be wrong, that has been known to happen, once.SIP trunks are supported in CM4.2 and CME 3.4 and 4.0See the data sheet for CM4.2 here:http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_data_sheet0900aecd8042402c.htmlYou're probably thinking of SIP endpoint functionality, which was addedin CM5.--Phil
		
		

		

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Re: [asterisk-users] Are you using app_meetme or app_conference

[Alyed Tzompa]

		Be careful when using heavily ChanSpy. We
did couple of weeks ago and the result was having Asterisk crashing
almost once every day. How heavy? around 4 people using it 8 hours a
day, each one using ChanSpy every 3-5 mins.
we were not able to find the exact reason, so just stop using it.Alyed 
		
		
		
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		On 15:27, Wed 27 Sep 06, Erick Perez wrote: Hi, for call centers with voip phones and calls coming in via SIP and Zap, what app_ are people using to do:We use SIP and IAX2 and SCCP (chan_sccp). Zap is notpossible for us because we want to run it on OpenBSD and thezaptel is not ported to it yet. -conference -listening to conversation of agentsChanSpy works for this.  Is app_meetme or app_conference?We use app_conference. We have to since there's no timerzaptel stuff for OpenBSD.  Does app_meetme still suffers from the need to transcode to slin?I have no idea.-- Michiel van Baak[EMAIL PROTECTED]http://michiel.vanbaak.euGnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD"Why is it drug addicts and computer afficionados are both called users?"___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Spurious hangups on zaptel interface

[Alyed Tzompa]

		I'm experiencing the same problems, but
unfortunatelly haven't been able to associate them with any number
since they appear to be random. But maybe we can do a little research
about it, and hopefully find teh solution for both:
are your PSTN lines POTS or E1/T1? can you make a couple of calls and
post here the logs? would be nice if you can enable the full Asterisk
log for a single call and post that one.Alyed 
		
		
		
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We seem to be getting unexpected hangups on our * system, very
consistent when calling particular numbers that we can associate with a
clients phone system. These hangups generally occur when our call is
transferred within their system (to voicemail usually). 
I'm suspecting their may be some sort of "flash" (for lack of a better term) on the called side, but I can't verify this. 
the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through.
Has anyone else experienced this, or have any potential resolutions?
I've researched this quite a bit, but not turning up anything
particularly relevant.
I am using asterisk 1.2.9.1Barry D. HasslerPresidentHCST
2332 Grange Hall Road
Beavercreek, Ohio 45431-2345
http://www.hcst.net/

 
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Re: [asterisk-users] Spurious hangups on zaptel interface

[Alyed Tzompa]

		I'm curious... why will this work??
busydetect will just cut the line if there are 4 tones (les or more
depending the busycount param), and call progress will in fact try not
to cut the call due to false hangups.Alyed
		
		
		
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		Barry D. Hassler wrote: We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually).   I'm suspecting their may be some sort of "flash" (for lack of a better term) on the called side, but I can't verify this.   the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through.  Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant.  I am using asterisk 1.2.9.1Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] H323 IP phones

[Alyed Tzompa]

		Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual appearance, end-user feedback, any infowill be appreciated.thnx!Alyed

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[asterisk-users] fw: Uniden - TVUNIDEN_UIP300

[Alyed Tzompa]

		Hi guys!
		Can someone give advice on nice H323 IP phones brands?
		?? 
I'm looking for some H323 IP phones for a customer. Diving in the
internet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard 
about them. Can someone give feedback experince about it??, config
ease, sound quality, visual appearance, end-user feedback, any info
will be appreciated.
thnx!
		
Alyed
		

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Re: [asterisk-users] g729 and polycoms problem

[Alyed Tzompa]

		Sorry but I've ran out of ideas...Anyone else out there with a successful Polycom g729 pass through-only experience?Alyed
		
		
		

Return-Path: [EMAIL PROTECTED] Thu Sep 21 11:27:21 2006Received: from nz-out-0102.google.com [64.233.162.206] by maila11.webcontrolcenter.com with SMTP;Thu, 21 Sep 2006 11:27:21 -0700Received: by nz-out-0102.google.com with SMTP id z6so390195nzd
		didn't work :(Regards,SantiagoOn 9/20/06, Alyed Tzompa <[EMAIL PROTECTED]>wrote: Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it? hey... you have in your sip.conf configuration "canreinvite=no"... think this may be a problem: since Asterisk will always stay in the path of the RTPs, I think it might need to have the proper transcoder, as it does not, then the error arises... at least that's what I think :) set "canreinvite=yes" (or just comment it since that's the default) on both parties and try again. Let me know if it works. Alyed  Return-Path: <[EMAIL PROTECTED]>Wed Sep 20 12:38:41 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 20 Sep 2006 12:38:41 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) Still having the same problem. i modified the sip.cfg in order to make g729 the first choice: voice.codecPref.G711A="3" voice.codecPref.G729AB="1" voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2" voice.codecPref.IP_4000.G729AB=""/ Cheers, Santiago On 9/19/06, Alyed Tzompa wrote:  Make sure the codec used by the Polycom will be only g729 via the phone's  web interface, as far as I remember Polycom will try always to use ulaw or  alaw first unless it is configured to use only or as first choice the g729  codec.   Alyed     Return-Path: Tue  Sep 19 14:47:54 2006  Received: from digium-69-16-138-164.phx1.puregig.net  [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;  Tue, 19 Sep 2006 14:47:54 -0700  Received: from digium-69-16-138-164.phx1.puregig.net  (localhost [127.0.0.1])  by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4;   Hi, I'm experiencing some problems with polycom phones, asterisk and g729  codec.   As I understand, between polycom and polycom i can use g729 without  license at all as long as I'm using codec_g729.so module (i'm using  the Open Source Implementation (  http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/  )  because it's pure pass-thru and there's no transcoding).   My sip.conf has the following options:   [general]  disallow=all  allow=g729  allow=ulaw[voipuser]  type=friend  username=user  host=dynamic  callerid=user 202  [EMAIL PROTECTED]  secret=gbvVf423  canreinvite=no  insecure=yes  disallow=all  allow=g729so i force the voipuser to use g729 as main codec. The problem comes  when i try to connect to other polycom phone with the same config as  voipuser. The CLI shows the following:   Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible  codecs!   show modules doesnt show codec_g729.so but if i try to load it i get this:   Unable to load module codec_g729.so  Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module  'codec_g729.so' already existsAnyone had this issue?   If you need more information, feel fre to ask for it :)Thanks a lot!   Santiago  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] g729 and polycoms problem

[Alyed Tzompa]

		Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it?
hey... you have in your sip.conf configuration "canreinvite=no"...
think this may be a problem: since Asterisk will always stay in the
path of the RTPs, I think it might need to have the proper transcoder,
as it does not, then the error arises... at least that's what I think :)
set "canreinvite=yes" (or just comment it since that's the default) on both parties and try again.
Let me know if it works.
Alyed 
		
		
		
Return-Path: [EMAIL PROTECTED] Wed Sep 20 12:38:41 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;   Wed, 20 Sep 2006 12:38:41 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])
		Still having the same problem. i modified the sip.cfg in order to makeg729 the first choice:voice.codecPref.G711A="3" voice.codecPref.G729AB="1"voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2"voice.codecPref.IP_4000.G729AB=""/Cheers,SantiagoOn 9/19/06, Alyed Tzompa  wrote:  Make sure the codec used by the Polycom will be only g729 via the phone's web interface, as far as I remember Polycom will try always to use ulaw or alaw first unless it is configured to use only or as first choice the g729 codec. Alyed   Return-Path:  Tue Sep 19 14:47:54 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP;  Tue, 19 Sep 2006 14:47:54 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])  by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4; Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ) because it's pure pass-thru and there's no transcoding). My sip.conf has the following options: [general] disallow=all allow=g729 allow=ulaw [voipuser] type=friend username=user host=dynamic callerid=user 202 [EMAIL PROTECTED] secret=gbvVf423 canreinvite=no insecure=yes disallow=all allow=g729 so i force the voipuser to use g729 as main codec. The problem comes when i try to connect to other polycom phone with the same config as voipuser. The CLI shows the following: Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! show modules doesnt show codec_g729.so but if i try to load it i get this: Unable to load module codec_g729.so Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module 'codec_g729.so' already exists Anyone had this issue? If you need more information, feel fre to ask for it :) Thanks a lot! Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Cisco 7970 behind NAT

[Alyed Tzompa]

		Since the phone is the one behind a NAT,
and the registration is done only with SIP packages, setting or not the
"nat" is not an issue (ONLY for registration purposes). You can see
this since Asterisk is receiving the registration. Why is it denying
it?... wel,  that's something that will most likely has to do with
the registrationn parameters (user-passwd), but certainly not with the
network configuration.Alyed
		
		
		
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		Does anyone have this working? I have a Cisco 7970 with the 8-0-2-SR1S firmware loaded on it. I can get the phone to register with * just fine when I place my asterisk server on the same subnet and do no NAT. When I give my asterisk server a static public IP and put the phone behind a NAT to connect to the server registration fails. I turn on sip debugging and see that the phone is trying to register but it gets 401 Unauthorized. The same phone config is being used with only modifications to the IPs of the proxy and some NAT settings. I've adjusted NAT settings in two places (phone config and sip.conf).Example:sip.confchange "nat=never" to "nat=yes"Phone config:change0to1Does anyone have a similar setup with a 7970 behind NAT to an asterisk server that is not behind NAT? Any help or thoughts would be greatly appreciated.Jeremiah___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] Configuring Codecs

[Alyed Tzompa]

		We can help guy, but pls send detailed info regarding a specific problem. The more info you provide, the best we can help.
As for g729 codec instructions look at
http://www.digium.com/en/supportcenter/documentation/viewdocs/G729 
there you'll find detailed and updated information about it.
Alyed 
		
		
		
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		Good Day,I
am new to Asterisk and I need help in configuring codecs in Asterisk. I
also will need to buy the license for G 729 codec and need to put that
key in Asterisk. Can someone please provide the step by step
instructions for the codec configurations?Also I am having
problems forwarding calls to landline or cellular numbers. As soon as
the preson on the forwarded end picks up the call drops - what can
cause this and what is the solution?ThanksWyatt___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] g729 and polycoms problem

[Alyed Tzompa]

		Make sure the codec used by the Polycom
will be only g729 via the phone's web interface, as far as I remember
Polycom will try always to use ulaw or alaw first unless it is
configured to use only or as first choice the g729 codec.Alyed 
		
		
		
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		Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec.As I understand, between polycom and polycom i can use g729 withoutlicense at all as long as I'm using codec_g729.so module (i'm usingthe Open Source Implementation (http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ )because it's pure pass-thru and there's no transcoding).My sip.conf has the following options:[general]disallow=allallow=g729allow=ulaw[voipuser]type=friendusername=userhost=dynamiccallerid=user 202[EMAIL PROTECTED]secret=gbvVf423canreinvite=noinsecure=yesdisallow=allallow=g729so i force the voipuser to use g729 as main codec. The problem comeswhen i try to connect to other polycom phone with the same config asvoipuser. The CLI shows the following:Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs!show modules doesnt show codec_g729.so but if i try to load it i get this:Unable to load module codec_g729.soSep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module'codec_g729.so' already existsAnyone had this issue?If you need more information, feel fre to ask for it :)Thanks a lot!Santiago___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] channel.c: Avoided initial deadlock for '0x8de2dc0', 10 retries!

[Alyed Tzompa]

		Hi there!
I'm having lots of problems with an Asterisk used by a customer. 
Got hundreds (yes hundreds, about 3-4 per minute) of this messages
every hour:
WARNING[12685] channel.c: Avoided initial deadlock for '0x96dee78', 10 retries!
The hex number changes with every message. The warning number does so as well but less often, thrice or twice a day. 
This Asterisk cuts offs calls once or twice, every day and is makes the
customer run into serious problems since it is a call center.
Asteriski version is: 1.2.10 with zaptel 1.2.7 and libpri 1.2.3
2 PRI ISDN E1's and about 50 concurrent calls constantly
Any ideas?
Many thanx!Alyed
		

		

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Re: [asterisk-users] nat and qualify questions

[Alyed Tzompa]

		

from
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
		

		 qualify=xxx|no|yes


where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds.

If you turn on qualify in the configuration of a SIP device in sip.conf, Asterisk will send a SIP OPTIONS
command regularly to check that the device
is still online. If the
device does not answer within the configured (or default) period (in
ms) Asterisk considers the device off-line for future calls.

		

What happens if you use nat=yes is that Asterisk will consider the IP
for communicating with the SIP user agent (UA) as the IP from where the
SIP invite comes from instead of taking the one included in the SDP
message. Hence if you are using phones inside a LAN this 2 addresses
will be the same, but if your SIP UA is outside they will not.
Having all your phones set with nat=yes and qualify =yes, will not
affect the behaviour of your phones if your network is not really full,
but will be a bad and dirty way to do it :)Alyed
		
		
		
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		As
far as i know qualify=yes will increase you network traffic, this will
make asterisk to communicate with all sip friends every X seconds, not
sure the default value.On 8/1/06, 
BerkHolz, Steven [EMAIL PROTECTED] wrote:Are there any 
problems with always having nat=yes and qualify=yes? We just opened up 
our server to be accessible to SIP from the internet. (used to require 
VPN) I had to set the SIP 
setting for my test softphone to nat=yes and qualify=yes.This makes 
sense. Some of these phone 
will never leave our building.Some of these phone 
will come and go. (laptops) Is the any negatives 
to just have all phones set to nat=yes and qualify=yes?If not, why is it 
not the default?  Thank You,Steven 
BerkHolz- MCSA 
- MCSE -Manager of Information SystemsTESCO Group 
CompaniesFax. 248-836-5101www.TESCOGroup.comBoard member 
ofwww.glimasoutheast.org
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re: [asterisk-users] SIP and NAT

[Alyed Tzompa]

		Could you please explain what the network configuration you want to try? it would be really helpful.
you can be as simple as:  SIPphone-- internet -- NAT-- asterisk
or whatever your particular scenario is.Alyed 
		
		
		
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		Hello all. I'm having a little problem here with NAT, and I already read a lot of documentation on web, but I still cant understand how to get asterisk and "external (on internet)" sip clients connected.Could anybody give me a tip ?ThanksLincoln___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] G729 Softphone

[Alyed Tzompa]

		As far as I there is no free softphone that can handle G729 codec. So you will need a licenced one.
Have a call center working with eyebeam from counterpath (previously
known as Xten) for about a year with no problems. Don't know if it
supports 
the URL option, but I'm pretty sure it will. Anyway you can ask this directly to their tech support. Alyed
		
		
		
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		Daniel Salama a écrit : Looking for a SIP or IAX softphone for a call center application that  can do G729 codec. Any recommendations? Ideally it would do screen pops,  meaning that it will understand the URL option of the Dial command.Of course I'm a little biased, but I think MozPhone is well suited to call center application: it does natively support URL option of Dial or Queue command. It does not support G729 though, but speex will give you nice quality / low bandwith.Have a look at http://moziax.mozdev.org/ and please send feedback / comment / questions to MozPhone's mailing list at: http://moziax.mozdev.org/list.htmlThanks,Jean-Denis___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] So many configuration files!

[Alyed Tzompa]

		first: download the latest version 1.2.5 had some bugs and is already several months old.Depending on how you want your asterisk to behave will be the amount of files you'll need to mess with. Let's say you want a very basic installation with some SIP phones (hard or soft), then you'll have to deal with sip.conf and extensions.conf only so everything else is just vanity :DUsually you would like to have some voicemail, conference rooms, music on hold, pick up your neighbours extension, dial another asterisk or an IAX softphone, and PSTN access, then change some configs in voicemail.conf, meetme.conf, musiconhold.conf, features.conf, iax.conf and zapata.conf respectively.Want more action?Manage asterisk from an external application and mess with manager.conf,  change the way logs are being saved and CRMs with logger.conf and  the cdr_ *.conf files,  try some text to speach (TTS) with festival.confFeel like you are in the right track?try dealing with any ".c" file, recompile asterisk and make it behave just the way you always dream of (btw if it works you might want to share your new feature with all of us :) )Alyed 
		
		
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		I'm working with Asterisk 1.2.5 to get a working system.There are 50 Asterisk configuration files in /etc/asterisk.Are they _all_ called by Asterisk or are some only used in a #include?Is there any way to get a list of which ones Asterisk uses by default?There is only a single #include file and it doesn't even exist.I have only messed with 4 files so far.Are there any more I should be editing?Which ones could be safely ignored?So far the system is just SIP with Zaptel to be added next.The 4 files I have changed are:sip.confextensions.confextensions_additional.confvoicemail.confMy list of files in /etc/asterisk - sorted most recent last:[EMAIL PROTECTED] asterisk # ls -1trzapata.confvpb.conftelcordia-1.adsiskinny.confsip_notify.confrtp.confrpt.confres_odbc.confqueues.confprivacy.confphone.confoss.confosp.confmusiconhold.confmodules.confmodem.confmisdn.confmgcp.confmeetme.confmanager.conflogger.confindications.confiaxprov.confiax.conffestival.conffeatures.confextensions.aelextconfig.confenum.confdundi.confdnsmgr.confcodecs.confcdr_tds.confcdr_pgsql.confcdr_odbc.confcdr_manager.confcdr_custom.confcdr.confasterisk.confasterisk.adsialsa.confalarmreceiver.confagents.confadtranvofr.confadsi.confsip.confextensions.confextensions_additional.confvoicemail.conf-- Larry Alkoff N2LA - Austin TXUsing Thunderbird on Slackware Linux___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [asterisk-users] trouble with * and # infront of a phonenumber

[Alyed Tzompa]

		As of Asterisk 1.0.X a "#" was recognized as a pattern not as a digit, hence in order to use it at the begining of an extension you should use "_" before it. I guess this is still valid in 1.2.X versions.i.e: use _#31#0046011 in your extensions.confAlyed 
		
		
		

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I cant make call when using #31#0046011
		


		


		 

		
The call just disapper  and noting shows on CLI 
		


		


		 

		


		 

		
Using asterisk 1.2.9.1
		


		
I am using cubix IAX2 softphone 
		


		


		 

		
And I had the same problem when I used xlite sip softphone 
		


		


		 

		
My voip-provider is Rixtelecom 
		


		


		 

		
//Michael Nielsen 
		


		


		 

		


		 
		
		

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re: [asterisk-users] setting of volume

[Alyed Tzompa]

		like if some one speaks loud, he would get a low volume.I'm sorry, but this goes far beyond Asterisk (at least for the moment) :)Anyway you can still play with rxgain and txgain in zapata.conf, but this will increase/reduce the overall volume gains and can also affect echo perception.Alyed 
		
		
		

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Is possible to set an default volume on a PBX, so that all lines have the same volume, like if some one speaks loud, he would get a low volume.
		


		


		 

		


		 

		
//Michael 
		

		
		

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re: [asterisk-users] Incoming Call matching to peer

[Alyed Tzompa]

		You have a little confusion:
friend = can GENERATE and RECEIVE calls
peer = can only GENERATE calls
user = can only RECEIVE callsAlyed 
		
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		I have this in sip.conf:[ata1]username = ata1accountcode = ata1qualify = yessecret = footype = friendhost = dynamicfromdomain = ipt.gumby.comcontext = fromata_startqualify = yesWhen
a call comes in from this device, if I have type=peer, Asterisk doesn't
match it, but it does if type=friend. Why not? I thought peers made and
received calls, and friends only made calls. If that's the case, then
it should match against type=peer, and certainly not against
type=friend.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] E1 additional calling party number

[Alyed Tzompa]

		Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number (CallerID)into an "Additional calling party number". Guess it
refeers to  a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf but
have not been able to send them the data the way they are asking.
Anyone there has an idea on what config shall I try??
So far with my configs I'm able to send them the data in the "Display" part of the trace.
My configurations look like this:
sip.conf
[general]
trustrpid =yes
restrictcid=yes / no
[1007]
host=dynamic
type=friend
secret=1007
callerid=1007
zapata.conf
usercallerid=yes
hidecallerid=no
usecallingpres=yesAlyed
		

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RE: Re: [asterisk-users] Help with MusicOnHold!!!

[Alyed Tzompa]

		2 things might worth having a look:
a) set up in your zapata.conf:
    musiconhold=default
b) You say the asterisk version is 1.1, but 1.1 is developement
version, maybe was just a typo, but you should be using either a 1.0.X
or 1.2.X versionAlyed 
		
		
		
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		Hi,
 
Yes we have that installed.  Music on hold works fine when called in extensions.conf 
e.g. 
exten = 5000,2,MusicOnHold()
 
However when I put someone on hold the music does not play
I am using the Polycom Soundstation IP301 and X-lite phones.
 
Thanks
 
Julian
 Date: Fri, 7 Jul 2006 09:35:56 +0800 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help with MusicOnHold 
Do you have the "mpg123" utility in your system? By default, Asterisk  uses "mpg123" to play the mp3 files for music on hold.  -kokmeng.  Julian Varanini wrote:   Hi,   
 I am running asterisk 1.1.  When a client is placed on hold from the 
 x-lite or polycom phone, no hold music is heard.  I have   musicclass=default set up in sip.conf and default exists in 
 musiconhold.conf.  Has anyone had a similar experience? Any help would   be appreciated.     Thanks     Julian    ___ --Bandwidth and Colocation provided by Easynews.com --  asterisk-users mailing list To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-users   ___ --Bandwidth and Colocation provided by Easynews.com --  asterisk-users mailing list To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] additional calling party number

[Alyed Tzompa]

		Hi there!
I'm setting up an E1 with a new Telco and they are asking me to add the
extension number into an "Additional calling party number". Guess it
refeers to  a part of the E1 trace they are getting. I've been
playing around with the callerid and in zapata.conf and sip.conf but
have not been able to send them the data the way they are asking.
Anyone there has an idea on what config shall I try??
So far with my config I'm able to send them the data in the "Display" part of the trace.
sip.conf
[general]
trustrpid =yes
restrictcid=yes / no
[1007]
host=dynamic
type=friend
secret=1007
callerid=1007
zapata.conf
usercallerid=yes
hidecallerid=no
usecallingpres=yesAlyed

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Re: [Asterisk-Users] Problem: ringtones stop unexpectedly

[Alyed Tzompa]

		Have you tryed phoning a fixed line instead of a cell phone?is this giving the same result?I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result.Alyed 
		
		

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		I should've mentioned that before. I've tried doing that and it has noeffect. I've tried both upper and lower-case 'r's.I've also tried a workaround that I thought would work, but it doesn't:Answering the call and then using the playtones(ringing) command beforeconnecting to my cellphone. -Original Message-Date: Sat, 1 Apr 2006 19:59:46 +0100From: "Julian J. M." <[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedlywhen multiple channels are dialedTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Message-ID:<[EMAIL PROTECTED]>Content-Type: text/plain; charset=ISO-8859-1Try adding 'r' to the dial options. According to "show application dial":r - Indicate ringing to the calling party. Pass no audio to thecallingparty until the called channel has answered.exten = 3058472194,1,Dial(SIP/1035SIP/[EMAIL PROTECTED],50, r)Julian.On 4/1/06, Carlos A. Alfaro <[EMAIL PROTECTED]>wrote: Hello Everyone. I usually find my own solutions for problems but thistime, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the caller, but they suddenly stop as the call to my cellphone starts to make progress. This causes some people to hang up, despite the fact that the call is stillbeing connected. Callers who stay on the line are able to talk to me on either the sip extension or the cellphone once I pick up either one. I have tried a lot of workarounds like including a priority to answer the incoming call, invoke the playtones command before the dial command, but this doesn't seem to work either. Can anyone replicate the problem? HaveI ran into a bug? I have pasted as much info as I deemed relevant; pleaselet me know if I'm missing something. Thanks.--___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersEnd of Asterisk-Users Digest, Vol 21, Issue 2*___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] G729 codec problems

[Alyed Tzompa]

		I used g729 couple of times in the past and got the warning messages ONLY when I was trying to use more channels than the total amount of licenses I'd got.If you are sure you are using only one device that needs the license, I would suggest to check out how it is communicating with Asterisk. Also, if you have enough time try using the g729 with another soft / IP phone and see if you get the same result.Alyed ---I am not. I have one license and use i channel.It seems to detect the fact there are no more channels left and keepswarning me about it in case I want to use more.It is fine, but the warning is constant. All you see on Asteriskconsole is running warning message.RudolfOn 4/2/06, Kevin P. Fleming <[EMAIL PROTECTED]>wrote: RumaTech wrote:  And it keeps running like that. Call usually come through OK. If i try  to use "show g729" command, it shows that all codecs are in use. Well,  this is fine, I am using one, but I do not want to see those warnings.  Once is quite enough. Those continuos warnings make it impossible to se  any other asterisk output. How does one turns them off? You can't make them stop except by not trying to use more channels than you have licenses for. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Re: no audio

[Alyed Tzompa]

		That was a bug fixed in Asterisk version 1.2.3  recently version 1.2.6 was released, so don't worry you can try the latest one without timing fears :DAlyed 
		
		

Return-Path: [EMAIL PROTECTED] Sat Apr 01 15:42:39 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 1 Apr 2006 15:42:39 -0700
		Wierd timing - I'm struggling with exactly the same issue. My problemwas with ZAP - ZAP. The phones ring, but no audio. Turns out there'sa bug with the version I'm running. It has to do w/ the system date. When I changed my system date to 1-Jan-06, everything worked!! Here'swhat I found from another posting:this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349change your system date to an older value. everything will work again.I'm hoping the bug is fixed in a more recent release build, but Ihaven't tried yet.Yours,HughOn 4/1/06, Luis herrera <[EMAIL PROTECTED]>wrote: Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with phones outside my network. I call the extensions without a problem, it rings but when they answer I can't hear them and they can hear me. I set up in the SIP.CONF nat=yes I'm I missing any other setting or do I need a special switch that support asterisk. Thank you for your help. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Asterisk and LCR

[Alyed Tzompa]

		I use Portaone's PortaSIP for everything related to LCRAlyed
		
		
		
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		I'm wondering how you guys handle least cost routing within Asterisk.Basically, we have a few providers with, obviously, different rates  per route. Additionally, we have a number of clients who have DIDs  assigned to them (either pointing to a single SIP peer or to more  complex dialing rules, such as, ring multiple SIP peers, or ring SIP  peers sequentially, or overflow to external forwarded number).What we are trying to do is to ease the pain of Asterisk dialing  through the least cost route regardless of where the destination is.  If the call is to go through a provider, well, then dial the least  cost provider. If someone is dialing a number which happens to be one  of the DIDs assigned in our system, it shouldn't have to go out  through any provider but just be routed properly in the dialing plan.Also, if a client has defined a rule to forward calls to another  number, how can I let the dial plan to dial out that number while  taking advantage of the LCR in place?We looked at lcdc script but not quite sure is what we need or maybe  don't understand it completely to implement it correctly.Thanks,Waldo___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Dumb question - reaching the PSTN

[Alyed Tzompa]

		I may add a very nice configuration:
-  Use two (or more) Asterisks to create your own VoIP network
Very useful if you have broadband and several facilities spread out in distant geographical locations.Alyed 
		
		
		
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		Not dumb at all.  Seeing that most end users in the world are still onthe PSTN, you're going to want to attach to it at some point.Since this is Asterisk, you have options.  Lots of them.  And, usingcareful dialplan-ning you can mix and match to your heart's content.Here are some examples:1) Use a Digium or a non-digium card to connect to POTS, ISDN, PRI, etc.2) Use a "in and out" VoIP provider such as telasip, vonage, etc3) Use an "out only" VoIP provider such as voipjet (generally lowerper-minute rates)4) Use an "in only" VoIP provider (examples fail me...)5) Use an FXO card (as in #1) but connect it to another device ratherthan the telco.  E.g. a cell phone6) Use a full-blown GSM gateway7) EtcUsing the above options, I personally have configured Asterisk to:1) Interface to my T12) Use voipjet to place outbound calls3) When dialing a company cell phone, send it out a cellular deviceuntil they're all used up, then use the T1Bob McDowell-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of CharlesMarcusSent: Wednesday, March 29, 2006 4:38 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Dumb question - reaching the PSTNHi everyone,I am fairly new to the idea of VoIP, although I've been reading about itoff and on for the last few years. Now it is starting to look matureenough to consider implementing it, but there is one thing that Ihaven't been able to get a clear answer on...With Vonage, you are using the Vonage network - it is theirresponsibility to route your call to the endpoint, which is more thanlikely on the old fashined PSTN.If I install Asterisk, how do my calls actually get completed? How dothey get 'bridged' over to the PSTN?I attended a Seminar today hosted by Dynasis, and one of the issues wasVoIP. ShoreTel was there, and the said I had to have phone lines,whether they were POTS lines, chennels from a T-1, whatever, we stillhad to have phone lines.Now I'm confused.If I implement an Asterisk based system (yes, I'd be paying a consultantto help), will I still have to maintain phone lines and pay full pricefor Long Distance?Simple pointers to White Papers on this issue will be sufficient.Many thanks,-- Best regards,Charles___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-usersEMAIL PRIVELEGED  CONFIDENTIAL CLIENT COMMUNICATION   *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION ***This
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re: [Asterisk-Users] Asterisk Between PBX and FXS

[Alyed Tzompa]

		As I understand this, it's a problem of redirecting the call to the same FXS channel.
To replicate this behaviour in the Asterisk you could try the following in the extensions.conf:
(suppose your FXS channel is group 1 in zapata.conf)
exten = 100,1,Dial(Zap/g1/${EXTEN},20)
exten = 100,1,Hangup
		

exten = 200,1,Dial(Zap/g1/${EXTEN},20)
exten = 200,1,Hangup
		

Then you'll end up with 2 extensions using the same FXS channel (of course not at the same time).
Hope this is what you are looking for.
Alyed
		
		
		
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		Hi guys,I'm setting up asterisk to run with another pbx server. This pbx serversupport a feature that allows 2 extensions connect to the same FXS. No I putasterisk in the middle.Asterisk receives the call and dial to a SIP/peer.How the pbx installed support 2 extensions to one fxs... How can I figure outin asterisk which extension was dialed before the call came to asterisk?Does asterisk receive this information in some variable?${BRIDGEPEER}${CALLERID(dndi)}${BLINDTRANSFER}${BLINDTRANSFER}I tried the above variables without success.Thanks in advance.Fernando Lujan___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] kernel recompilation on a asterisk server

[Alyed Tzompa]

		Think a zaptel recompile is just what you need.Alyed 
		
		
		
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		i've got a asterisk server running on slackware 10.2 sice 3 weeks, iwill upgrade the kernel to 2.6.x from 2.4.x , i will upgrade to thelast stable release of kernel...i've got asterisk 1.2.4, zaptel 1.2.4 and libpri 1.2.2, correctlycompiled and configured, obviusly if i try to load the new kernel ican't load wtcxx modules, should i recompile zaptel 1.2.4 with make make install or there is a specific procedure to follow?thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] [OT] Polycom provisioning

[Alyed Tzompa]

		Polycom's can work in one of two ways:
a) using self configuration
b) downloading it from a ftp server
To make your Polycoms work with Asterisk you actually don't need the
phone to download any configuration, with the one embeded is ok. In any
case, when turned on, the phone searches for the ftp server, when it
cannot find it, it just continues loading with its self pre-configured
parameters.
Once this process is finished you should keep on with your own local network and sip configurationsAlyed
		
		
		
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		Does anyone have the polycom soundpoint ip's successfully remotely provisioning?  I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information.  Any help would be appreciated :)-- Aaron DanielComputer Systems TechnicianSam Houston State University[EMAIL PROTECTED](936) 294-4198___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] User Extension Custom Voicemail

[Alyed Tzompa]

		if you want to use the defaults unavailable and bussy just put it in /var/spool/asterisk/voicemail/default/100/unavail.wav
The "100" extension will automatically be created after you leave your
first voicemail message, change "unavail.wav" for "busy.wav" to use
them as
Voicemail(u100) and 
		Voicemail(b100) respectivelly in your extensions.conf
		
		
Alyed 
		
		
		
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		I created extension 100 in my asterisk system for myself.  I then wentahead and created a WAV file to use for my voicemail message.  Where wouldI go about putting this WAV file so that I can use it?Thank you,Jyran GluckyAdvisory ProgrammerBlueWare, Inc.Strategic HealthWare Solutions3060 W. 13th StreetCadillac, MI 49601Phone:  (231) 779-0224 ext. 111Fax: 231-779-1002Skype: Jyran GluckyAIM: JyranGluckymailto:[EMAIL PROTECTED]http://www.blueware.netDID YOU KNOW?BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2(Document Management) Application Worldwide.BlueWare Market Share for Hospital Document Management Systems is in 25states in the US.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] hardware and network requirements

[Alyed Tzompa]

Have a customer running some 25-28 concurrents calls (with about 35 agents logged in)without problems with a P4 2.X Ghz, 1GB RAM,I'm doing no transcoding btw.Alyed  Return-Path: [EMAIL PROTECTED] Sat Feb 04 16:59:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 4 Feb 2006 16:59:29 -0700 i'm planning to migrate a callcenter to asterisk and VOIP,  the call center can have up to 25 cuncurrents agents logged in. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000One of our clients has a similar sized setup running on an Athlon64 2800+(2.2Ghz I think), 1GB RAM, 2x80GB HDDs in RAID1.You don't say how the calls are coming in, but I'd try and keep transcodingto a minimum. if they're coming from a PRI (i.e. alaw or ulaw) and you wantto keep them that way down to the users, 25 concurrent calls @ 80kbps-ish isonly 2mbps, so even a 100mbps LAN is fine for the task.Personally, I build our asterisk boxes rather than buying off-the-shelfservers, but I doubt it makes much difference one way or t'other. Go withwhichever approach you feel most comfortable.Regards,Chris-- C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] nwebmail

[Alyed Tzompa]

Also get the book (again I dont have the URL if some one does please post it).  http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Alyed   If you are new I would reccomend using [EMAIL PROTECTED]http://asteriskathome.soundforge.net . It is a greatresource for beginers. Also get the book (again I donthave the URL if some one does please post it).AsteriskRegards,Dovid--- yrving rivas  wrote: Hello!  I am new to Asterisk, AMP, Linux...did I say all?.. I just installed Asterisk, and for my needs it is working great. In my AMP I see the nwebmail but I cant get into it. When I place my login and password, comes with the following message: "An internal error has occured. Please contact your system administrator. If you are the system administrator, check the log files".  The log files dont help me very much.  Can someone tell me how to use the nwebmail?, how to get in for first time?  Regards!  Yrving- Do You Yahoo!? La mejor conexin a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --Bandwidth and Colocation provided by Easynews.com --  Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Problem with an automatic responder

[Alyed Tzompa]

I would be useful if you could post your config files and the pri debug as well.  check your zapata.conf or paste it here so we can take a look.AlyedReturn-Path: [EMAIL PROTECTED] Thu Jan 12 10:04:28 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 12 Jan 2006 10:04:28 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 6DAB2C258; Thu, 12 Jan 2006 09:58:40 -0700 (MST)Received: from psmtp.com (exprod5mx128.postini.com [64.18.0.42]) by lists.digium.com (Postfix) with SMTP id 4101AC24B for asterisk-users@lists.digium.com; Thu, 12 Jan 2006 09:58:34 -0700 (MST)Received: from source ([151.9.129.69]) (using TLSv1) by exprod5mx128.postini.com ([64.18.4.10]) with SMTP;  Thu, 12 Jan 2006 10:58:41 CSTReceived: from aulin (aulin.pitagora.it [193.227.67.249]) by allserv.pitagora.it (8.12.11/8.12.11) with ESMTP id k0CGwXKd006897 for asterisk-users@lists.digium.com; Thu, 12 Jan 2006 17:58:33 +0100Received: from VIGGIANI ([193.227.66.207]) by MERCURIO.pitagora.it with Microsoft SMTPSVC(6.0.3790.1830); Thu, 12 Jan 2006 17:58:32 +0100X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comFrom: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Thu, 12 Jan 2006 17:58:32 +0100Message-ID: [EMAIL PROTECTED]MIME-Version: 1.0Content-Type: text/plain; charset="us-ascii"Content-Transfer-Encoding: 7bitX-Mailer: Microsoft Office Outlook 11Thread-Index: AcYXmWqjqLUMCcjoQT2Cz9B43oUplQ==X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2180X-OriginalArrivalTime: 12 Jan 2006 16:58:32.0672 (UTC) FILETIME=[6B723600:01C61799]X-Spam-Status: No, score=-4.8 required=5.5 tests=ALL_TRUSTED,AWL,BAYES_00  autolearn=ham version=3.0.4X-Spam-Checker-Version: SpamAssassin 3.0.4 (2005-06-05) on allserv.pitagora.itX-Virus-Scanned: ClamAV version 0.88, clamav-milter version 0.87 on localhostX-Virus-Status: CleanX-pstn-levels: (S: 4.16697/99.78420 )X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1X-pstn-addresses: from [EMAIL PROTECTED] [90/4]Subject: [Asterisk-Users] Problem with an automatic responderX-BeenThere: asterisk-users@lists.digium.comX-Mailman-Version: 2.1.5Precedence: listReply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comList-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.comList-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED]List-Archive: http://lists.digium.com/pipermail/asterisk-usersList-Post: mailto:asterisk-users@lists.digium.comList-Help: mailto:[EMAIL PROTECTED]List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED]Sender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: [EMAIL PROTECTED]Hi,I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phonesand a few of VoIP phones directly connected to Asterisk.Calling a number (only one until now!) - an automatic responder (IVR) - fromVoIP phones works, from analog phones doesn't work: NOANSWER after a fewseconds. I'm using no 'r' in dial options (this caused a problem with an IVRsome time ago).I can post pri debug output in both cases, if needed.Thanks in advance for any help-- Mimmus___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] read .what else to do ?

[Alyed Tzompa]

Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP.  just add the range ports tih a ":" e.g 192.168.1.2 1 : 10007(4)Please,I know alot of you out there have implemented AAH to work  outside your network ( Setting up your router/firewall so your remote SIP  phones can communicate with your [EMAIL PROTECTED] Server via SIP through a  NAT ).Please advise me how to make it work !!!  If what you are trying to do is a SIP -- NAT -- Internet -- Nat -- Asterisk call them I'm afraid you would need to use a SIP/RTP router. Alyed Return-Path: [EMAIL PROTECTED] Thu Jan 12 09:29:42 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Please note that recent IOS has SIP NAT traversal turned on by default.I believe that it only supports internal UA / external server.Since you also want the opposite, you should probably turn it off:no ip nat service sip tcp port 5060no ip nat service sip udp port 5060Some IOS versions will even crash on SIP behind NAT. Seehttp://lists.digium.com/pipermail/asterisk-users/2004-January/033718.htmlSorry, I don't know how to forward a range of ports. To forwarda single port, use something like:ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendablewhere x.x.x.x is your public IP.You can edit rtp.conf to use e.g 1-10007 (would allow 4 calls) andthen only 8 ip nat statements would be needed for RTP.You don't say what's failing. "make calls outside our LAN" sounds likeyou are trying to call using a VoIP provider that Asterisk registerswith. But "your remote SIP phones" is something different; which ofthe above are failing? Are the registrations successful? Is it justthe RTP that's not working (in which case the called phone will stillring)? If not, what error or timeout is reported?If * verbose and/or debug logs don't show precisely what is going wrong,use Ethereal (on both sides of the router if necessary) to see whatis happening.--Stewart Hi all , I have tried configuring Asterisk at home to make calls outside our Lan WITHOUT any success (Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT )  To be precise i did the following  (1) I Forwarded UDP Port 5060-5082 to 192.168.1.2 Forward UDP Port 1 to 2 to 192.168.1.2  (2) I set externip = x.x.x.x (to our public WAN) localnet =192.168.1.0 /255.255.255.0  (3) I also set nat=yes qualify=yes  (4)Please,I know alot of you out there have implemented AAH to work outside your network ( Setting up your router/firewall so your remote SIP phones can communicate with your [EMAIL PROTECTED] Server via SIP through a NAT ).Please advise me how to make it work !!!  (5) I am using xten lite soft phone on my pc .  (6) I use cisco 1700 series router ,and i have natting configured on this router .Maybe I am using a wrong command .Please,tell me the commands to forward the ports Port 5060-5082,1 to 2 to 192.168.1.2 on a cisco router .  Please reply and advice !!! Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] No D-channels available! Using Primary on channel 16 anyway!

[Alyed Tzompa]

I had a very similar problem some months ago, was using a Sangoma A101 card though. The problem was something related to the card's memory and was able to solve it by updating the driver. It was caused due to I was using a brand new card with a not so updated driver (I was using one that I thought was "stable") So my advice here is to check the driver version you are using if not the very last one, then update it. Try looking at the /var/log/messages file for any extra info, you might find something interesting. Alyed Return-Path: [EMAIL PROTECTED] Thu Jan 12 06:16:56 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Thu, 12 Jan 2006 06:16:56 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 6834A4066; Thu, 12 Jan 2006 06:15:45 -0700 (MST)Received: from psmtp.com (exprod5mx148.postini.com [64.18.0.180]) by lists.digium.com (Postfix) with SMTP id ACB784060 for asterisk-users@lists.digium.com;Hi!I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks on span 3 and 4 and * 1.2.Every few hours I get this message and asterisk dies just after that:Warning: No D-channels available! Using Primary on channel  16 anyway!When this happens restarting zaptel and asterisk services, generally puts the system back onlinemy zaptel.con reads:span=1,1,0,ccs,hdb3 #span=2,0,0,cas,hdb3 span=3,2,0,esf,b8zs #-- This because we have two American CBsspan=4,3,0,esf,b8zsbchan=1-15dchan=16bchan=17-31fxoks=63-86fxoks=87-110loadzone = usIdeas anybody? Please?  Things done:* zttool/ztcfg* Trying R2 instead of PRI (R2 is the south americanstatdar, which wont even start)*Added crc4 to span1, with ugly sound consequences-- Paavum Regina, Per Secula et Secularum!!
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re: [Asterisk-Users] Outbound routing

[Alyed Tzompa]

can't you ask the users to dial a prefix? that can solve your problem. btw, which provider are you using for your calls to the USA? Alyed Return-Path: [EMAIL PROTECTED] Wed Jan 11 09:47:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Wed, 11 Jan 2006 09:47:29 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 91BB3C4BD; Wed, 11 Jan 2006 09:17:16 -0700 (MST)Received: from psmtp.com (exprod5mx130.postini.com [64.18.0.44]) by lists.digium.com (Postfix) with SMTP id C7900C473 for asterisk-users@lists.digium.com; Wed, 11 Jan 2006 09:17:04 -0700 (MST)Received: from source ([200.93.220.22]) (using TLSv1) by exprod5mx130.postini.com ([64.18.4.10]) with SMTP;  Wed, 11 Jan 2006 08:17:09 PSTReceived: from ozzy (srv.eth0.www.manta.telconet.net [200.93.220.18]) by smtp.manta.telconet.net (8.12.11/8.12.11) with ESMTP id k0BGPPla027509 for asterisk-users@lists.digium.com; Wed, 11 Jan 2006 11:25:25 -0500X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comFrom: Guillermo Salas M [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comContent-Type: text/plainOrganization: Telconet S.A.Date: Wed, 11 Jan 2006 11:16:55 -0500Message-Id: [EMAIL PROTECTED]Mime-Version: 1.0X-Mailer: Evolution 2.2.3Content-Transfer-Encoding: 7bitX-Spam-Status: No, score=-4.1 required=7.5 tests=ALL_TRUSTED,AWL,BAYES_00, DNS_FROM_RFC_ABUSE autolearn=unavailable version=3.1.0X-Spam-Checker-Version: SpamAssassin 3.1.0 (2005-09-13) on  smtp.manta.telconet.netX-Virus-Scanned: ClamAV 0.87/1237/Tue Jan 10 10:53:20 2006 on smtp.manta.telconet.netX-Virus-Status: CleanX-pstn-levels: (S: 8.29151/99.85851 )X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1X-pstn-addresses: from [EMAIL PROTECTED] [90/4]Subject: [Asterisk-Users] Outbound routingX-BeenThere: asterisk-users@lists.digium.comX-Mailman-Version: 2.1.5Precedence: listReply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comList-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.comList-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED]List-Archive: http://lists.digium.com/pipermail/asterisk-usersList-Post: mailto:asterisk-users@lists.digium.comList-Help: mailto:[EMAIL PROTECTED]List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED]Sender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: [EMAIL PROTECTED]Hi all, I've 3 providers (A, B, and C) the A is giving me freecalls toUSA, the B is giving my freecalls to Europe, and C is to call the otredestinations. My question is, how can I configure the outboud routing toselect the right trunk for every destination?All the providers uses the dialing form 00 1 123 4567890 when 00 is thenumber dialed to call, 1 the country code, 123 the area code and 4567890the phone number.I've the following outbound routing with AMP, but the calls are beenstarted by the first provider in the trunk sequence list:Route Name : InternationalDial Patterns : 00.Trunk Sequence: A B CI want to make that the USA calls going with A, Europe calls with B andrest of the world with C.Is this possible ? Can you gime a little of help with this... Than you in advance. :)-- Guillermo Salas M.Telconet S.A. MantaCalle 15 y Av. 24 Esq.Phone : 593 5 262 8071Mobile: 593 9 985 5138SIP : [EMAIL PROTECTED]e-mail: [EMAIL PROTECTED]www : http://www.telconet.net http://www.telcocarrier.netLinux User: 255902Soporte en Linea en http://www.manta.telconet.netPlease avoid sending me Word or PowerPoint attachments.See http://www.fsf.org/philosophy/no-word-attachments.html___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] how to adjust volume

[Alyed Tzompa]

Don't know if you can actually adjust the volume in any of them, but you can try from the asterisk with rxgain / txgain in your zapata.confAlyed Return-Path: [EMAIL PROTECTED] Mon Jan 09 16:27:24 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Mon, 9 Jan 2006 16:27:24 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id BAA93C839; Mon, 9 Jan 2006 16:26:29 -0700 (MST)Received: from psmtp.com (exprod5mx147.postini.com [64.18.0.179]) by lists.digium.com (Postfix) with SMTP id 08788C829 for asterisk-users@lists.digium.com; Mon, 9 Jan 2006 16:26:26 -0700 (MST)Received: from source ([64.233.162.206]) by exprod5mx147.postini.com ([64.18.4.10]) with SMTP; Mon, 09 Jan 2006 15:26:29 PSTReceived: by zproxy.gmail.com with SMTP id o1so4491534nzf for asterisk-users@lists.digium.com; Mon, 09 Jan 2006 15:26:29 -0800 (PST)Received: by 10.36.221.48 with SMTP id t48mr13251723nzg; Mon, 09 Jan 2006 15:26:29 -0800 (PST)Received: by 10.36.97.3 with HTTP; Mon, 9 Jan 2006 15:26:29 -0800 (PST)X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comDate: Mon, 9 Jan 2006 23:26:29 +From: Asterisk guy [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comPrecedence: listReply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: [EMAIL PROTECTED]how to adjust voice volume for sipura 2000 and cisco ata186?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Problem with integrating ISDN PBX using NT mode

[Alyed Tzompa]

 == Primary D-Channel on span 4 down for TEI 65 == Primary D-Channel on span 4 down for TEI 64 == Primary D-Channel on span 4 down for TEI 66 This looks like a signaling problem, check out your configuration in zaptel.conf and the log messages you get at /var/log/messages You can also post them here if you continue experiencing problems.   Alyed Hi,I'm just in the process of replacing a crappy Siemens PBX with a new andshiny Asterisk system. To connect Legacy equipment I hooked up a smallISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRIcard. That port is configured for NT Point to Multipoint(Mehrgeraeteanschluss) mode. Now I can place calls from the ISDN PBX tothe other Asterisk extensions but the other way around does not work.Whenever I call from the Asterisk server to one of the extensionsconnected through the ISDN PBX that extension rings for a split secondand then the call is dropped. Here is what I get on the console: -- Executing Macro("SCCP/13-002f", "standard|Zap/g2/40") in new stack -- Executing Dial("SCCP/13-002f", "Zap/g2/40|20") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/40 == Primary D-Channel on span 4 up for TEI 64 == Primary D-Channel on span 4 up for TEI 66 -- Zap/10-1 is proceeding passing it to SCCP/13-002f -- Zap/10-1 is ringing -- Channel 0/1, span 4 got hangup request -- Hungup 'Zap/10-1' == No one is available to answer at this time (1:0/0/0) -- Executing Goto("SCCP/13-002f", "s-NOANSWER|1") in new stack -- Goto (macro-standard,s-NOANSWER,1) -- Executing VoiceMail("SCCP/13-002f", "u40") in new stack -- Executing Goto("SCCP/13-002f", "default|s|1") in new stack -- Goto (default,s,1) == Channel 'SCCP/13-002f' jumping out of macro 'standard' == Primary D-Channel on span 4 down for TEI 65 == Primary D-Channel on span 4 down for TEI 64 == Primary D-Channel on span 4 down for TEI 66I think I properly configured the ISDN PBX (theres not much to configurethere). Can someone help me here? What's causing the hangup request? Howcould I find out?Below is the relevant configuration.Thanks in advance,Frederik Fixzapata.conf:[channels]switchtype = euroisdnpridialplan = localprilocaldialplan = localnationalprefix = 0internationalprefix = 00;usecallingpres=yesechocancel = yesechocancelwhenbridged = yesechotraining = 100debug = 2; Festnetzanschlusssignalling = bri_cpecontext=externgroup = 1; S/T port 1channel = 1-2; S/T port 2channel = 4-5; S/T port 3channel = 7-8; Interner S0-Bussignalling = bri_net_ptmpcontext = intern-isdngroup = 2; S/T port 4channel = 10-11extensions.conf:[macro-standard]exten = s,1,Dial(${ARG1},20)exten = s,2,Goto(s-${DIALSTATUS},1)exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})exten = s-NOANSWER,2,Goto(default,s,1)exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})exten = s-BUSY,2,Goto(default,s,1)exten = _s-.,1,Goto(s-NOANSWER,1)exten = a,1,VoicemailMain(${MACRO_EXTEN})[nebenstellen-intern]; Konferenzzimmerexten = 13,1,Macro(standard,SCCP/13); Ingridexten = 17,1,Macro(standard,${INGRID})exten = 57,1,Macro(standard_ohne_ab,Zap/g2/57); Gavigoexten = 60,1,Macro(standard,SCCP/60); Woelmexten = 66,1,Macro(standard,SCCP/66)exten = 68,1,Macro(standard_ohne_ab,Zap/g2/68); van de Beeckexten = 40,1,Macro(standard,Zap/g2/40)exten = 44,1,Macro(standard_ohne_ab,Zap/g2/44); Rohanexten = 50,1,Macro(standard,Zap/g2/50)exten = 59,1,Macro(standard_ohne_ab,Zap/g2/59); Hinterhausexten = 58,1,Macro(standard,Zap/g2/58)exten = 22,1,Macro(standard,Zap/g2/22); fuer Testzweckeexten = 61,1,Macro(standard,SIP/eyebeamtest); virtuelle Nebenstellenexten = 30,1,Macro(virtuell)exten = 35,1,Macro(virtuell)exten = 48,1,Macro(virtuell)exten = 25,1,Dial(Zap/g2/58)[intern-isdn]exten = 25,1,Dial(SCCP/13)exten = s,1,DISA(no-password|intern)[dialout]exten = _0.,1,Dial(Zap/g1/${EXTEN:1})[intern]include = dialoutinclude = nebenstellen-intern___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] Asterisk Debugging

[Alyed Tzompa]

I don't find the console output ugly, maybe messy, but never ugly :P  If u don't like those NoOp, just take them away from ur extensions.conf. BTW, to save the console output to a given file, just edit your logger.conf file.  Say you only want the console output, then just add to your filename the verbose option . The file will be saved wherever is defined in the asterisk.conf (the default is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI logger rotate at the Asterisk console. i.e.  ;logger.conf  [logfiles] mylogfile = verbose Alyed I'd like to have Asterisk log useful messages during operation.Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway.Thanks,Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Asterisk Debugging

[Alyed Tzompa]

Then stop looking for easy solutions and get your hands dirty changing your c files Alyed  Well, I want the output that the NoOp's generate. I want to be able to manually log lines to a file through some mechanism. I just wish I could do it without all the extra NoOp stuff at the front.  I just tried using: mylogfile = verbose  in logger.conf but all I got was the startup/shutdown asterisk messages. Besides, this isn't what I wan't. I don't want Asterisk internal generated log messages. I want my OWN log messages, that I specify.  Doug-Original Message-From: Alyed Tzompa [mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006 11:18 AMTo: asterisk-users@lists.digium.comSubject: re: [Asterisk-Users] Asterisk DebuggingI don't find the console output ugly, maybe messy, but never ugly :PIf u don't like those NoOp, just take them away from ur extensions.conf. BTW, to save the console output to a given file, just edit your logger.conf file. Say you only want the console output, then just add to your filename the verbose option . The file will be saved wherever is defined in the asterisk.conf (thedefault is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI logger rotate at the Asterisk console.i.e. ;logger.conf[logfiles]mylogfile = verboseAlyed   I'd like to have Asterisk log useful messages during operation.Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway.Thanks,Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Asterisk Debugging

[Alyed Tzompa]

I do agree, plus even if you don't know anything about scripting there are plenty of shell tutorials out thereAlyed No but shell scripts are pretty easy and will cleanup your file for you.On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote: Not everyone is a C programmer extraordinairre. -Original Message- From: Alyed Tzompa [mailto:[EMAIL PROTECTED] Sent: Thursday, January 05, 2006 11:59 AM To: Douglas Garstang; asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk Debugging Then stop looking for easy solutions and get your hands dirty  changing your c files___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] confusion about contexts - SER

[Alyed Tzompa]

Guess I got where your confusiion lies you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf  So what you have to do is the following:  -user 2092, set it the createmenu context in sip .conf - in extensions.conf under the "createmenu" context add as many "include" lines as you need the user to have acces to.  it should look something like: ; sip.conf  [2092] type=friend username=2092 canreinvite=no context= createmenu  ; extensions.conf [createmenu] ... ... include = outgoing include = some other context   hope this helps AlyedHi,  Thanks for the reply.  What happens is that all users are registered with SER (a sip proxy). I have set SER up so when a user dials 0 followed by a pstn number it will be forwarded to asterisk which will forward the call to a third party pstn gateway. I also use asterisk so that when a user who is registered with ser doesnt answer (sending a 408 cancel response) or is busy (sending a 486 busy response) that the call is forwarded to asterisk voicemail. So therefore at the moment I have a user 2092 which registers with ser and uses the outgoing context in asterisk for pstn access and accesses their voicemail mailbox through the default context.  Now I also set it up so that if a user registered with SER dials 20005 it should forwards to asterisk. This should call the context createmenu which creates an IVR menu.  What Im confused about is this. I created a user 20005 in sip.conf using context=createmenu. This wasnt working. After reading your post I realized my mistake was that the context that is being called is that of the caller i.e. 2092 as opposed to whom the call is directed at i.e. 20005. Therefore when I changed the context of 2092 to createmenu it worked.  BUT how can I set up my sip.conf so that 2092 can use the default, outgoing and createmenu contexts depending on the correct scenario? If someone who is also using SER has any comments, Id also really appreciate it.  i.e.  [300] type=friend username=300 canreinvite=no context= WHAT GOES HERE?? //createmenu calls the IVR but then outgoing pstn calls dont work, outgoing allows pstn calls but then I cant create a menu etc etc insecure=very ;callerid= "voicemail user 1" 300 host=dynamic nat=yes dtmfmode=INFO mailbox=300 disallow=all allow=alaw allow=ulaw allow=g723.1 allow=g729  Many thanks, Aisling.   -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alyed Tzompa Sent: 04 January 2006 00:28 To: asterisk-users@lists.digium.com Subject: re: [Asterisk-Users] confusion about contexts  I'm a bit confused on how you get your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip user? in this  case, which one?, if not is it iax or though a zap channel?  anyway, here some tips:  For your first problem it seems it has to do with what I pointed above, check that the user which is dialing into asterisk has  the correct context (context=create-menu) with at least type= peer  also don't have to retype the allow=codec, disallow=codec, dtmfmode=x for every user, just set it in the general context in  sip.conf  your second problem think it has to do once again with the firts thing above, and regarding the retyping, I'm afaid I don't know  any other way than writing those lines again and again for everyuser. Maybe someone else out thereknows someting else that can help.  Don't set many "outgoing" context for every user in sip.conf! just set one and point all users to that one. If you need your  user to have acces to other contextsjust add  include = your_context at the end of whatever context you want (btw can add more than oneinlcude's)   Alyed  --- Hi,  Hope someone can help me-Asterisk isnt behaving as I would expect and I think its down to my contexts.  There are two things I cant fathom.  Firstly I want to record an IVR and so have created a user 20005 and a context called createmenu. I am using SER in front of asterisk so I changed the ser.cfg so that if the user dialled this number it forwards to asterisk. This works fine. The problem is when the invite reaches my asterisk box, asterisk uses the wrong context. It appears to call the outgoing context which is the context used to route calls to my pstn gateway provider. It then trys to execute a Dial command for 20005 which isnt supposed to happen.  Secondly SER uses Asterisk for voicemail if a phone doesnt answer after a certain period of time or is busy. This works fine but I have to create an entry for every user in extensions.conf under the [default] context. Can I create a generic entry which would also work to shorten the config file?...Also if I change this and out all the entries under a context voicemail it doesnt work.I have to keep it in defaultTh

Re: [Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

made the changes in sip.conf so now it reads:  disallow=all allow ilbc  now I when the call is placed it is not hanged up, but I cannot hear anything. I think it's becasue Asterisk is sending the RTP's to a wrong address (my internal IP). Looked at the sip debug and got the following:   -- Executing BackGround("SIP/alyed-5a8d", "/var/lib/asterisk/sounds/testt") in new stack We're at 200.78.243.12 port 13458 Answering with preferred capability 0x400(ILBC) Answering with non-codec capability 0x1(G723) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 90.0.0.10;branch=z9hG4bK5a0a00c043bab4f9390f1bef02ef;received=201.127.53.246;rport=5060 From: "unknown"sip:[EMAIL PROTECTED]:5060;tag=2438130825771721203 To: sip:[EMAIL PROTECTED]:5060;tag=as7222f729 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 220  v=0 o=root 17028 17028 IN IP4 200.78.243.12 s=session c=IN IP4 200.78.243.12 t=0 0 m=audio 13458 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  to 201.127.53.246:5060  -- Playing '/var/lib/asterisk/sounds/test' (language 'en') Integra2*CLI  Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 90.0.0.10;rport;branch=z9hG4bK5a0a00c043bab4f944b4f6f302f2 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK From: "unknown"sip:[EMAIL PROTECTED]:5060;tag=2438130825771721203 Max-Forwards: 70 To: sip:[EMAIL PROTECTED]:5060;tag=as7222f729 User-Agent: SJphone/1.60.299a/L (SJ Labs)   9 headers, 0 linesany ideas?Return-Path: [EMAIL PROTECTED] Mon Jan 02 22:32:10 2006Received: from bourbon.fnords.org [209.16.72.158] by mail11.webcontrolcenter.com with SMTP; Mon, 2 Jan 2006 22:32:10 -0700Received: from [172.18.3.242] (24-179-48-91.static.slid.la.charter.com [24.179.48.91]) (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits)) (No client certificate requested) by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88; Mon, 2 Jan 2006 23:32:08 -0600 (CST)Message-ID: [EMAIL PROTECTED]Date: Mon, 02 Jan 2006 23:30:25 -0600From: "Eric \"ManxPower\" Wieling" [EMAIL PROTECTED]User-Agent: Thunderbird 1.5 (Windows/20051201)MIME-Version: 1.0To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] SIP through freeBSD NATReferences: [EMAIL PROTECTED]In-Reply-To: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1; format=flowedContent-Transfer-Encoding: 7bitX-SmarterMail-Spam: SPF_NoneAlyed Tzompa wrote: sip.conf [general] port=5060 externip = www.theip.net localnet = 192.168.1.0 localmask = 255.255.255.0 allow=allDon't use allow=all. Use disallow=all and then allow= line for the specific codec you want to use.
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Re: [Asterisk-Users] Asterisk PRI problems.

[Alyed Tzompa]

It should be possible, but I think that the problem lies in the connection between Avaya and Asterisk, let me explain: your Avaya most likely uses different connection parametrs for handling the calls coming from /going to PSTN and the ones coming from /going to Asterisk.   If you need to pass let's say extensions' 20XX traffic, from Avaya through Asterisk, just set this in the Avaya and let it dial to Asterisk as if it was another extension, once in the Asterisk, you can handle it with the extensions.conf Alyed Return-Path: [EMAIL PROTECTED] Tue Jan 03 00:22:49 2006Received: from mailfront1.citynet.nu [217.10.96.36] by mail11.webcontrolcenter.com with SMTP; Tue, 3 Jan 2006 00:22:49 -0700Received: from localhost (localhost [127.0.0.1]) by mailfront1.citynet.nu (Postfix) with ESMTP id 76B104C0A0; Tue, 3 Jan 2006 08:22:47 +0100 (CET)Received: from mailfront1.citynet.nu ([127.0.0.1]) by localhost (mailfront1.citynet.nu [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id 12166-02; Tue, 3 Jan 2006 08:22:46 +0100 (CET)Received: from localhost (unknown [10.1.2.65]) by mailfront1.citynet.nu (Postfix) with ESMTP id 3D43F4C08A; Tue, 3 Jan 2006 08:22:45 +0100 (CET)Date: Tue, 3 Jan 2006 10:19:45 +0100From: Kristian Larsson [EMAIL PROTECTED]To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] Asterisk PRI problems.Message-ID: [EMAIL PROTECTED]References: [EMAIL PROTECTED]Mime-Version: 1.0Content-Type: text/plain; charset=iso-8859-15Content-Disposition: inlineIn-Reply-To: [EMAIL PROTECTED]User-Agent: Mutt/1.5.8iX-Virus-Scanned: by amavis at citynet.nuX-Spam-Status: No, hits=-4.5 tagged_above=-999.0 required=999.0 tests=ALL_TRUSTED, BAYES_00X-Spam-Level: X-SmarterMail-Spam: SPF_NoneOn Mon, Jan 02, 2006 at 12:09:33PM -0700, Alyed Tzompa wrote: You are not gonna be able to modify this behaviour from the asterisk since in your case asterisk is only receiving the digits from someone else (an Avaya in your case but could be PSTN for instance)  Just asked an Avaya support guy and told me you should take a look at the ARS Digit Analysis Table, and modify the Min and Max values to suit your needs.I just found something called "digit delay time"and "digit delay count", it just happens so thatdigit delay count is set to four, ie the number ofdigits before the avaya dials out. I can not tryit out right now since the system is being used -has to wait for the maintenance window tomorrownevening.But what I find a little strange is that the Avayaworks when connected to my telco, shouldn't it bepossibly to simulate a telco fully with Asterisk?Thanks Alyed! Kristian. Hope this helps ww6   Return-Path:  Mon Jan 02 05:56:31 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Mon, 2 Jan 2006 05:56:31 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id C0C434092; Mon, 2 Jan 2006 05:55:56 -0700 (MST) Received: from psmtp.com (exprod5mx128.postini.com [64.18.0.42]) by lists.digium.com (Postfix) with SMTP id 20B414082 for ; Mon, 2 Jan 2006 05:55:52 -0700 (MST) Received: from source ([217.10.96.36]) by exprod5mx128.postini.com ([64.18.4.10]) with SMTP; Mon, 02 Jan 2006 06:55:53 CST Received: from localhost (localhost [127.0.0.1]) by mailfront1.citynet.nu (Postfix) with ESMTP id 892034C08A for ; Mon, 2 Jan 2006 13:55:53 +0100 (CET) Received: from mailfront1.citynet.nu ([127.0.0.1]) by localhost (mailfront1.citynet.nu [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id 17711-06 for ; Mon, 2 Jan 2006 13:55:53 +0100 (CET) Received: from localhost (unknown [10.1.2.65]) by mailfront1.citynet.nu (Postfix) with ESMTP id 236B64C06A for ; Mon, 2 Jan 2006 13:55:52 +0100 (CET) X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com Date: Mon, 2 Jan 2006 15:52:51 +0100 From: Kristian Larsson  To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Mime-Version: 1.0 Content-Type: text/plain; charset=iso-8859-15 Content-Disposition: inline User-Agent: Mutt/1.5.8i X-Virus-Scanned: by amavis at citynet.nu X-Spam-Status: No, hits=-3.8 tagged_above=-999.0 required=999.0 tests=ALL_TRUSTED, BAYES_05 X-Spam-Level:  X-pstn-levels: (S:65.75150/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from  [90/4] Subject: [Asterisk-Users] Asterisk PRI problems. X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion  List-Id: Asterisk Users Mailing List - Non-Commercial Discussion  List-Unsubscribe: ,  List-Archive:  List-Post:  List-Help:  List-Subscribe: ,  Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] X-SmarterMail-Spam: SPF_None X-Rcpt-To:I have an Avaya IP Office PBX connected to an Asterisk system via a S

Re: [Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

SJphone supports ilbc, anyway tryed it with ulaw, alaw and gsm (all of them supported by SJphone), but the behaviour is the same. That's why I thought this sould be a RTP addressing stuff Alyed   Return-Path: [EMAIL PROTECTED] Tue Jan 03 11:46:59 2006Received: from bourbon.fnords.org [209.16.72.158] by mail11.webcontrolcenter.com with SMTP; Tue, 3 Jan 2006 11:46:59 -0700Received: from [172.16.13.73] (unknown [172.16.13.73]) (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits)) (No client certificate requested) by bourbon.fnords.org (Postfix) with ESMTP id 91F6386; Tue, 3 Jan 2006 12:46:58 -0600 (CST)Message-ID: [EMAIL PROTECTED]Date: Tue, 03 Jan 2006 12:45:15 -0600From: "Eric \"ManxPower\" Wieling" [EMAIL PROTECTED]User-Agent: Thunderbird 1.5 (Windows/20051201)MIME-Version: 1.0To: [EMAIL PROTECTED]Cc: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] SIP through freeBSD NATReferences: [EMAIL PROTECTED]In-Reply-To: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1; format=flowedContent-Transfer-Encoding: 7bitX-SmarterMail-Spam: SPF_NoneUse a codec your phone supports like ulaw.Alyed Tzompa wrote: made the changes in sip.conf so now it reads:  disallow=all allow ilbc  now I when the call is placed it is not hanged up, but I cannot hear  anything. I think it's becasue Asterisk is sending the RTP's to a wrong  address (my internal IP). Looked at the sip debug and got the following:  -- Executing BackGround("SIP/alyed-5a8d",  "/var/lib/asterisk/sounds/testt") in new stack We're at 200.78.243.12 port 13458 Answering with preferred capability 0x400(ILBC) Answering with non-codec capability 0x1(G723) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP  90.0.0.10;branch=z9hG4bK5a0a00c043bab4f9390f1bef02ef;received=201.127.53.246;rport=5060 From: "unknown";tag=2438130825771721203 To: ;tag=as7222f729 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:  Content-Type: application/sdp Content-Length: 220  v=0 o=root 17028 17028 IN IP4 200.78.243.12 s=session c=IN IP4 200.78.243.12 t=0 0 m=audio 13458 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  to 201.127.53.246:5060 -- Playing '/var/lib/asterisk/sounds/test' (language 'en') Integra2*CLI  Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP  90.0.0.10;rport;branch=z9hG4bK5a0a00c043bab4f944b4f6f302f2 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK From: "unknown";tag=2438130825771721203 Max-Forwards: 70 To: ;tag=as7222f729 User-Agent: SJphone/1.60.299a/L (SJ Labs)   9 headers, 0 linesany ideas? Return-Path:  Mon Jan 02 22:32:10 2006 Received: from bourbon.fnords.org [209.16.72.158] by  mail11.webcontrolcenter.com with SMTP; Mon, 2 Jan 2006 22:32:10 -0700 Received: from [172.18.3.242] (24-179-48-91.static.slid.la.charter.com  [24.179.48.91]) (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits)) (No client certificate requested) by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88; Mon, 2 Jan 2006 23:32:08 -0600 (CST) Message-ID: [EMAIL PROTECTED] Date: Mon, 02 Jan 2006 23:30:25 -0600 From: "Eric \"ManxPower\" Wieling"  User-Agent: Thunderbird 1.5 (Windows/20051201) MIME-Version: 1.0 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion   Subject: Re: [Asterisk-Users] SIP through freeBSD NAT References:  In-Reply-To:  Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-SmarterMail-Spam: SPF_None  Alyed Tzompa wrote:  sip.conf  [general]  port=5060  externip = www.theip.net  localnet = 192.168.1.0  localmask = 255.255.255.0  allow=all  Don't use allow=all. Use disallow=all and then allow= line for the specific codec you want to use. 
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Re: [Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

Tryed what Eric suggested in the other thread (changing in sip.conf: allow=all for   disallow=all allow=somecodec) so now the call is not being hanged up, but cannot hear anything. Tryied it with ilbc,alaw, ulaw and gsm  I still think it sould be a matter of RTP addressing since I get the following after a sip debug :   -- Executing BackGround("SIP/alyed-5a8d", "/var/lib/asterisk/sounds/test") in new stack We're at 200.78.243.12 port 13458 Answering with preferred capability 0x400(ILBC) Answering with non-codec capability 0x1(G723) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 90.0.0.10;branch=z9hG4bK5a0a00c043bab4f9390f1bef02ef;received=201.127.53.246;rport=5060 From: "unknown"sip:[EMAIL PROTECTED]:5060;tag=2438130825771721203 To: sip:[EMAIL PROTECTED]:5060;tag=as7222f729 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 220  v=0 o=root 17028 17028 IN IP4 200.78.243.12 s=session c=IN IP4 200.78.243.12 t=0 0 m=audio 13458 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -  to 201.127.53.246:5060  -- Playing '/var/lib/asterisk/sounds/test' (language 'en') Integra2*CLI  Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 90.0.0.10;rport;branch=z9hG4bK5a0a00c043bab4f944b4f6f302f2 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK From: "unknown"sip:[EMAIL PROTECTED]:5060;tag=2438130825771721203 Max-Forwards: 70 To: sip:[EMAIL PROTECTED]:5060;tag=as7222f729 User-Agent: SJphone/1.60.299a/L (SJ Labs)   9 headers, 0 lines   think it is addressing the rtp to my internal IP, but don't know who can I address it in the right way. I'm using the default STUN config in the SJphone : STUN server address -- stun.softjoys.com :3478, refresh time out --120 conclusive response timeout--0 retrunsmissions number -- 13  and nat= yes in the sip.conf  But still no sound in my endpoint Alyed  Alyed Tzompa SosaSimitelVoIP developer+52 (55) 24 52 43 90 Ext. (107)[EMAIL PROTECTED]Return-Path: [EMAIL PROTECTED] Mon Jan 02 13:29:11 2006Received: from mail08.syd.optusnet.com.au [211.29.132.189] by mail11.webcontrolcenter.com with SMTP; Mon, 2 Jan 2006 13:29:11 -0700Received: from router.vpn.marvy.net (c220-237-85-239.kelvn1.qld.optusnet.com.au [220.237.85.239]) by mail08.syd.optusnet.com.au (8.12.11/8.12.11) with ESMTP id k02KT9sT019578 (version=TLSv1/SSLv3 cipher=EDH-RSA-DES-CBC3-SHA bits=168 verify=NO) for [EMAIL PROTECTED]; Tue, 3 Jan 2006 07:29:09 +1100Received: from alcan (alcan.vpn.cc.uq.edu.au [192.168.136.2]) by router.vpn.marvy.net (8.13.3/8.13.3) with SMTP id k02KT88l022322 for [EMAIL PROTECTED]; Tue, 3 Jan 2006 06:29:08 +1000 (EST) (envelope-from [EMAIL PROTECTED])Message-ID: [EMAIL PROTECTED]From: "Tony Jago" [EMAIL PROTECTED]To: [EMAIL PROTECTED]References: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] SIP through freeBSD NATDate: Tue, 3 Jan 2006 06:29:27 +1000MIME-Version: 1.0Content-Type: multipart/alternative; boundary="=_NextPart_000_0016_01C6102F.0BCE3AD0"X-Priority: 3X-MSMail-Priority: NormalX-Mailer: Microsoft Outlook Express 6.00.2900.2670X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2670X-SmarterMail-Spam: SPF_None "Unable to agree on media streams" means that the two devices can't pick a common codec. This shouldn't have anything to do with the firewall.  - Original Message -  From: Alyed Tzompa  To: asterisk-users@lists.digium.com  Sent: Tuesday, January 03, 2006 5:32 AM Subject: [Asterisk-Users] SIP through freeBSD NAT Hi everyoneMy problem is the following:I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT(a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk.My soft phone registers without problems to the Asterisk but once the record starts to play I get a hangup. SJphone outputs"End reason: Unable to agree on media streams". I'm forwarding SIP and IAX ports from the remote NAT towards the Asterisk box (i've tryied it with IAX with no problems) usingthe following config in the remote NAT:/etc/pf.conf.# outgoing UDP port 5060 connections use source port 5060 on firewallnat on $ext_if inet proto udp from any port = 5060 to any - ($ext_if) port 5060# Redirect all trafic from NAT:asterisk_port to 192.168.1.5:asterisk_portrdr on $ext_if inet proto { tcp, udp } from any to any port 4569 - 192.168.1.5 port 4569rdr on $ext_if inet proto { tcp, udp } from any to $ext_if port 5060 - 192.168.1.5 port 5060rdr on $ext_if inet proto { tcp , udp} from any to any port 1:2 - 192.168.1.5 port 1:2# Let the Internet see our services pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port 4569 keep state pass in log-a

Re: [Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

it does support ilbc, alaw, ulaw and gsm. I've tryied all but get the same results with all of them the phone doesn't hangs up, but cannot hear anything in my endpoint. Alyed Return-Path: [EMAIL PROTECTED] Tue Jan 03 12:47:02 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Tue, 3 Jan 2006 12:47:02 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])Eric "ManxPower" Wieling wrote: Use a codec your phone supports like ulaw. Assuming he is using SJphone, that I understand, would support iLBC even in the free version ? Alyed Tzompa wrote:  made the changes in sip.conf so now it reads: disallow=all allow ilbc now I when the call is placed it is not hanged up, but I cannot hear  anything. I think it's becasue Asterisk is sending the RTP's to a  wrong address (my internal IP). Looked at the sip debug and got the following: -- Executing BackGround("SIP/alyed-5a8d",  "/var/lib/asterisk/sounds/testt") in new stack We're at 200.78.243.12 port 13458 Answering with preferred capability 0x400(ILBC) Answering with non-codec capability 0x1(G723) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP  90.0.0.10;branch=z9hG4bK5a0a00c043bab4f9390f1bef02ef;received=201.127.53.246;rport=5060  From: "unknown";tag=2438130825771721203 To: ;tag=as7222f729 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:  Content-Type: application/sdp Content-Length: 220 v=0 o=root 17028 17028 IN IP4 200.78.243.12 s=session c=IN IP4 200.78.243.12 t=0 0 m=audio 13458 RTP/AVP 97 101 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 201.127.53.246:5060 -- Playing '/var/lib/asterisk/sounds/test' (language 'en') Integra2*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP  90.0.0.10;rport;branch=z9hG4bK5a0a00c043bab4f944b4f6f302f2 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK From: "unknown";tag=2438130825771721203 Max-Forwards: 70 To: ;tag=as7222f729 User-Agent: SJphone/1.60.299a/L (SJ Labs) 9 headers, 0 lines any ideas?  Return-Path:  Mon Jan 02 22:32:10 2006 Received: from bourbon.fnords.org [209.16.72.158] by  mail11.webcontrolcenter.com with SMTP; Mon, 2 Jan 2006 22:32:10 -0700 Received: from [172.18.3.242] (24-179-48-91.static.slid.la.charter.com  [24.179.48.91]) (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits)) (No client certificate requested) by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88; Mon, 2 Jan 2006 23:32:08 -0600 (CST) Message-ID: [EMAIL PROTECTED] Date: Mon, 02 Jan 2006 23:30:25 -0600 From: "Eric \"ManxPower\" Wieling"  User-Agent: Thunderbird 1.5 (Windows/20051201) MIME-Version: 1.0 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion   Subject: Re: [Asterisk-Users] SIP through freeBSD NAT References:  In-Reply-To:  Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-SmarterMail-Spam: SPF_None Alyed Tzompa wrote:  sip.conf  [general]  port=5060  externip = www.theip.net  localnet = 192.168.1.0  localmask = 255.255.255.0  allow=all Don't use allow=all. Use disallow=all and then allow= line for the specific codec you want to use.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] How do you check whether a channel is active and the number of calls

[Alyed Tzompa]

Just type in the asterisk command line: show channels or sip show channels  type "help" also to take a look at the other commands availableAlyed How do you check whether a channel is active and the number of calls on it?Is it simple and complicated?/ObelixThis message was sent using IMP, the Internet Messaging Program.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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re: [Asterisk-Users] confusion about contexts

[Alyed Tzompa]

I'm a bit confused on how you get your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip user? in this case, which one?, if not is it iax or though a zap channel?anyway, here some tips:For your first problem it seems it has to do with what I pointed above, check that the user which is dialing into asterisk has the correct context (context=create-menu) with at least type= peeralso don't have to retype the allow=codec, disallow=codec, dtmfmode=x for every user, just set it in the general context in sip.confyour second problem think it has to do once again with the firts thing above, and regarding the retyping, I'm afaid I don't know any other way than writing those lines again and again for everyuser. Maybe someone else out thereknows someting else that can help.Don't set many "outgoing" context for every user in sip.conf! just set one and point all users to that one. If you need your user to have acces to other contextsjust add include = your_contextat the end of whatever context you want (btw can add more than oneinlcude's)Alyed ---Hi,Hope someone can help me-Asterisk isnt behaving as I would expectand I think its down to my contexts.There are two things I cant fathom.Firstly I want to record an IVR and so have created a user 20005 anda context called createmenu. I am using SER in front of asterisk so Ichanged the ser.cfg so that if the user dialled this number itforwards to asterisk. This works fine. The problem is when the invitereaches my asterisk box, asterisk uses the wrong context. It appearsto call the outgoing context which is the context used to routecalls to my pstn gateway provider. It then trys to execute a Dialcommand for 20005 which isnt supposed to happen.Secondly SER uses Asterisk for voicemail if a phone doesnt answerafter a certain period of time or is busy. This works fine but I haveto create an entry for every user in extensions.conf under the[default] context. Can I create a generic entry which would also workto shorten the config file?...Also if I change this and out all theentries under a context voicemail it doesnt work.I have to keepit in defaultThis must obviously be something got to do withAsterisk finding the contexts.I am confused as to how you apply multiple contexts to one user. Atthe moment nearly each user (besides 20005 and 1234) has a context ofoutgoing in sip.conf. This is so that they can make outgoing pstncallsBut what if I needed them to use another context in othersituations?...Im just confused as to what context should be applied.I have included the relevant parts of my sip.conf and extensions.confbelow. I would appreciate any advice as to why these issues areoccurring.Many thanks,Aisling.;sip.conf[general]bindport=5064bindaddr=0.0.0.0disallow=allallow=ulawallow=alawallow=gsmsrvlookup=yescanreinvite=no;autocreatepeer=yesnat=yesdtmfmode=infoinsecure=veryregisterattempts=0register = username:[EMAIL PROTECTED]/1234;To receive incoming calls specify this and replace"yourcontext-pstn" for your dial plan[blueface-in]type=peerhost=sip.blueface.iecontext=pstn[1234]type=friendusername=1234canreinvite=nocontext=pstninsecure=very;callerid= "Ais" 1234host=dynamicnat=yesdtmfmode=INFOmailbox=1234disallow=allallow=alawallow=ulawallow=g723.1allow=g729;added below line(s) for BLUEFACE conf;To make outgoing calls specify this block[blueface-out]type=peerhost=sip.blueface.ieusername=usernamesecret=password[20005]type=friendusername=20005canreinvite=nocontext=createmenuinsecure=very;callerid= "Ais" 20005host=dynamicnat=yesdtmfmode=INFOmailbox=20005disallow=allallow=alawallow=ulawallow=g723.1allow=g729[300]type=friendusername=300canreinvite=nocontext=outgoinginsecure=very;callerid= "voicemail user 1" 300host=dynamicnat=yesdtmfmode=INFOmailbox=300disallow=allallow=alawallow=ulawallow=g723.1allow=g729extensions.conf[general]static=yeswriteprotect = yes[createmenu]exten = 20005,1,Wait(2)exten = 20005,2,Record(/tmp/asterisk-recording:gsm)exten = 20005,3,Wait(2)exten = 20005,4,Playback(/tmp/asterisk-recording)exten = 20005,5,wait92)exten = 20005,6,Hangup;specify context for receiving incoming calls[pstn];Note this is just an example there are infinite different ways tohandle the incoming call.;exten = 1234, 1,Wait(1);exten = 1234, 2,Playback(beep);exten = 1234, 3,Hangupexten = 1234, 1, Dial(SIP/[EMAIL PROTECTED]) ; 1234 is the contact extension, defaultcontact extension is "s";exten = 2092,1,Answer();exten = 2092,2,Playback(welcome);exten = 2092,3,Background(menu);exten = 1,1,Dial($316);exten = 2,1,Dial($314)[outgoing]; Dial the Blue Face Speaking Clockexten = 300,1,Dial(SIP/[EMAIL PROTECTED])exten = 300,2,Hangup;Send PSTN calls to Blue Faceexten = _X.,1,Dial(SIP/[EMAIL PROTECTED])exten = _X.,2,Hangup[default]exten = 300, 1,Dial(SIP/300,20)exten = 300, 2,Voicemail(u300)exten = 300, 102,Voicemail(b300)exten = 300, 103,Hangupexten = 301, 1,Dial(SIP/301,20)exten = 301, 2,Voicemail(u301)exten = 301, 102,Voicemail(b301)exten = 

re: [Asterisk-Users] Asterisk PRI problems.

[Alyed Tzompa]

You are not gonna be able to modify this behaviour from the asterisk since in your case asterisk is only receiving the digits from someone else (an Avaya in your case but could be PSTN for instance)Just asked an Avaya support guy and told me you should take a look at the ARS Digit Analysis Table, and modify the Min and Max values to suit your needs.Hope this helpsww6  Return-Path: [EMAIL PROTECTED] Mon Jan 02 05:56:31 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Mon, 2 Jan 2006 05:56:31 -0700Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id C0C434092;Mon, 2 Jan 2006 05:55:56 -0700 (MST)Received: from psmtp.com (exprod5mx128.postini.com [64.18.0.42])by lists.digium.com (Postfix) with SMTP id 20B414082for asterisk-users@lists.digium.com;Mon, 2 Jan 2006 05:55:52 -0700 (MST)Received: from source ([217.10.96.36]) by exprod5mx128.postini.com([64.18.4.10]) with SMTP; Mon, 02 Jan 2006 06:55:53 CSTReceived: from localhost (localhost [127.0.0.1])by mailfront1.citynet.nu (Postfix) with ESMTP id 892034C08Afor asterisk-users@lists.digium.com;Mon, 2 Jan 2006 13:55:53 +0100 (CET)Received: from mailfront1.citynet.nu ([127.0.0.1])by localhost (mailfront1.citynet.nu [127.0.0.1]) (amavisd-new,port 10024)with ESMTP id 17711-06 for asterisk-users@lists.digium.com;Mon, 2 Jan 2006 13:55:53 +0100 (CET)Received: from localhost (unknown [10.1.2.65])by mailfront1.citynet.nu (Postfix) with ESMTP id 236B64C06Afor asterisk-users@lists.digium.com;Mon, 2 Jan 2006 13:55:52 +0100 (CET)X-Original-To: asterisk-users@lists.digium.comDelivered-To: asterisk-users@lists.digium.comDate: Mon, 2 Jan 2006 15:52:51 +0100From: Kristian Larsson [EMAIL PROTECTED]To: asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Mime-Version: 1.0Content-Type: text/plain; charset=iso-8859-15Content-Disposition: inlineUser-Agent: Mutt/1.5.8iX-Virus-Scanned: by amavis at citynet.nuX-Spam-Status: No, hits=-3.8 tagged_above=-999.0 required=999.0tests=ALL_TRUSTED, BAYES_05X-Spam-Level: X-pstn-levels: (S:65.75150/99.9 )X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1X-pstn-addresses: from [EMAIL PROTECTED] [90/4]Subject: [Asterisk-Users] Asterisk PRI problems.X-BeenThere: asterisk-users@lists.digium.comX-Mailman-Version: 2.1.5Precedence: listReply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comList-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.comList-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED]List-Archive: http://lists.digium.com/pipermail/asterisk-usersList-Post: mailto:asterisk-users@lists.digium.comList-Help: mailto:[EMAIL PROTECTED]List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED]Sender: [EMAIL PROTECTED]Errors-To: [EMAIL PROTECTED]X-SmarterMail-Spam: SPF_NoneX-Rcpt-To: [EMAIL PROTECTED]I have an Avaya IP Office PBX connected to anAsterisk system via a Sangoma ISDN PRI card.Dialing from the asterisk system into the avaya works just fine butwhen trying to call from a phone connected to theavaya system something goes wrong. After punching the firstfour digits the Avaya calls out, shouldn't it waitfor all digits and then dial out?If I try to dial a three digit number it waits fora while then dials.Is this some feature to let the CO know of whicharea code the calls is going ahead of time?Is there some way to circumvent this using hackson the asterisk side?-- Kristian Larsson, Net At Once ABEmail: [EMAIL PROTECTED]Phone: +46 470 592717Cell: +46 704 910401___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] SIP through freeBSD NAT

[Alyed Tzompa]

Hi everyone My problem is the following: I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT(a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk. My soft phone registers without problems to the Asterisk but once the record starts to play I get a hangup. SJphone outputs"End reason: Unable to agree on media streams".  I'm forwarding SIP and IAX ports from the remote NAT towards the Asterisk box (i've tryied it with IAX with no problems) usingthe following config in the remote NAT: /etc/pf.conf . # outgoing UDP port 5060 connections use source port 5060 on firewall nat on $ext_if inet proto udp from any port = 5060 to any - ($ext_if) port 5060 # Redirect all trafic from NAT:asterisk_port to 192.168.1.5:asterisk_port rdr on $ext_if inet proto { tcp, udp } from any to any port 4569 - 192.168.1.5 port 4569 rdr on $ext_if inet proto { tcp, udp } from any to $ext_if port 5060 - 192.168.1.5 port 5060 rdr on $ext_if inet proto { tcp , udp} from any to any port 1:2 - 192.168.1.5 port 1:2 # Let the Internet see our services pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port 4569 keep state  pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port 5060 keep state  . -- I think the problem might relay in this "pass in log-all" since once I commented the last line and the SJphone was unable toregister, but I haven't found a way to set up a range using this "pass" command (it complains saying that the " : " is valid onlywith the "rdr " command) but I haven't found info explaining why I should (or shouldn't) use this "pass" command. My Asterisk config is: sip.conf [general] port=5060 externip = www.theip.net localnet = 192.168.1.0 localmask = 255.255.255.0 allow=all [user] nat=yes canreinvite=no host=dynamic extensions.conf exten = 400,1,Background(/var/lib/asterisk/sounds/myrecord) exten = 400,2,Hangup exten = 400,102,Hangup - Thanx a lot! ww6 
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[Asterisk-Users] sip through nat problem

[Alyed Tzompa]

Hi everyone  My problem is the following: I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT (a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk.  My soft phone registers without problems to the Asterisk but once the record starts to play I get a hangup. SJphone outputs "End reason: Unable to agree on media streams".   I'm forwarding SIP and IAX ports from the remote NAT towards the Asterisk box (i've tryied it with IAX with no problems) using the following config in the remote NAT:  /etc/pf.conf . # outgoing UDP port 5060 connections use source port 5060 on firewall nat on $ext_if inet proto udp from any port = 5060 to any - ($ext_if) port 5060  # Redirect all trafic from NAT:asterisk_port to 192.168.1.5:asterisk_port rdr on $ext_if inet proto { tcp, udp } from any to any port 4569 - 192.168.1.5 port 4569 rdr on $ext_if inet proto { tcp, udp } from any to $ext_if port 5060 - 192.168.1.5 port 5060 rdr on $ext_if inet proto { tcp , udp} from any to any port 1:2 - 192.168.1.5 port 1:2  # Let the Internet see our services pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port 4569 keep state pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port 5060 keep state . --  I think the problem might relay in this "pass in log-all" since once I commented the last line and the SJphone was unable to register, but I haven't found a way to set up a range using this "pass" command (it complains saying that the " : " is valid only with the "rdr " command) but I haven't found info explaining why I should (or shouldn't) use this "pass" command.   My Asterisk config is:  sip.conf [general] port=5060 externip = www.theip.net localnet = 192.168.1.0 localmask = 255.255.255.0 allow=all  [user] nat=yes canreinvite=no host=dynamic extensions.conf  exten = 400,1,Background(/var/lib/asterisk/sounds/myrecord) exten = 400,2,Hangup exten = 400,102,Hangup -   Thanx a lot! ww6
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