[asterisk-users] cmd Authenticate

[Coco Richard]

Hi,

i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?

thx
rich

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Re: [asterisk-users] cmd Authenticate

[Coco Richard]

Danny, Doug

thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
use Read().

rich

On Tue, Jun 29, 2010 at 3:26 PM, Doug Lytle supp...@drdos.info wrote:
 Coco Richard wrote:
 Hi,

 i need to save into a local variable the user's input dialed during
 the cmd Authenticate(). Is there a way to do it?



 core show application authenticate
 hylafax*CLI
   -= Info about application 'Authenticate' =-

 [Synopsis]
 Authenticate a user


   Options:
      a - Set the channels' account code to the password that is entered

 --

 You probably could use option a.

 But, I'd suggest that instead of using authenticate, you code something
 using the read option.

 I use read to authenticate conference administration.

 [check-password]

 exten = s,1,Read(get-admin-password|enter-password|||3|)
 exten = s,n,Gotoif($[${LEN(${get-admin-password})}  1]?9:3)
 exten = s,n, some dialplan magic here.

 Doug


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Re: [asterisk-users] Security Against brute force attack

[Coco Richard]

Hi,

there are several possibilities do to it

REGISTER Username/Extensions Enumeration
INVITE Username/Extensions Enumeration
OPTION Username/Extensions Enumeration

for more information:
http://www.hackingvoip.com/presentations/sample_chapter3_hacking_voip.pdf

rich...


On Thu, Nov 19, 2009 at 12:46 AM, Rasmus Männa aster...@razu.pri.ee wrote:

  Hi All,

 I must say that there are many ways to detect password attack cause this
 information actually goes into logs and it's possible to analyze them.
 Couple of hours thinking + day or 2 creating gives a really nice result. Bad
 thing is that by the time someone will start guessing password with
 dictionary attack or brute force (it doesn't matter) he already knows what
 is the account name/ID.

 All this leads me to question which is (from my point of view) a bit more
 important. Is there any way to detect SIP/IAX account guessing without
 actually dumping UDP flow ? I tried some _hacking_ tools and these create
 only some logs in debug mode. Using debug is not always an option cause in
 some cases it creates ~5MB log in a minute - such flow is quite impossible
 to handle.

 Does anyone have any experience catching account guessing attempts
 automatically ? Any kind of ideas would be wonderful :)

 thx a lot,
 --
 razu


 On 11/18/2009 10:01 PM, Ioan Indreias wrote:

 Hello Xavier,

  Unfortunately we are not aware of any Asterisk configuration which will
 protect against of a brute force attack on SIP.

  We use BFD - http://www.rfxn.com/projects/brute-force-detection/ .

  We have found first details here: http://engineertim.com/?cat=15 and
 we are currently maintaining 4 rules (SIP and IAX) . All of them could be
 downloaded from here:
 http://www.modulo.ro/Modulo/downloads/tools/tenora.bfd.tar.gz

  We have tried to document the installation of BFD on an Asterisk server
 here:
 http://www.modulo.ro/Modulo/ro/Articole/Securitate_pentru_servere_Asterisk.html
  (in
 Romanian)


  HTH,
 Ioan (Nini) Indreias
 www.modulo.ro


 On Mon, Nov 16, 2009 at 7:24 PM, TDF aja101...@gmail.com wrote:

 fail2ban


 http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk


 2009/11/16 Xavier Mesquida xavi...@yahoo.com

   Has Asterisk any protection against brute force attack for SIP
 authentication?
 Something like a maximum login attempt limit
 Thanks




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Re: [asterisk-users] Allow Header

[Coco Richard]

Hi,

asterisk version is 1.4.13

rich...

On Tue, Nov 10, 2009 at 7:01 AM, Tilghman Lesher tles...@digium.com wrote:
 On Monday 09 November 2009 15:38:54 Coco Richard wrote:
 i'm not sure to understand. Asterisk does support SIP INFO, so why
 doesn't Asterisk add the INFO Method in the 200OK Response?

 You must be using Asterisk 1.2.  This is the only version that I could find
 that does not put the INFO tag into the Allow header.  Asterisk 1.4 and all
 versions greater supply the INFO tag as standard.

 Given that 1.2 is in security-only fix mode now, this is not going to be
 changed in SVN or in any subsequent 1.2 release (if any).  You're welcome to
 change the ALLOWED_METHODS define in the top of chan_sip.c and
 recompile, however.

 --
 Tilghman Lesher
 Digium, Inc. | Senior Software Developer
 twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
 Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] Allow Header

[Coco Richard]

I took a look in chan_sip.c an for 1.4.13 ALLOWED_METHODS doesn't add
INFO. So I will upgrade to 1.6...

thank you for the replies...

rich...


On Tue, Nov 10, 2009 at 9:21 AM, Coco Richard
richard.kingc...@gmail.com wrote:
 Hi,

 asterisk version is 1.4.13

 rich...

 On Tue, Nov 10, 2009 at 7:01 AM, Tilghman Lesher tles...@digium.com wrote:
 On Monday 09 November 2009 15:38:54 Coco Richard wrote:
 i'm not sure to understand. Asterisk does support SIP INFO, so why
 doesn't Asterisk add the INFO Method in the 200OK Response?

 You must be using Asterisk 1.2.  This is the only version that I could find
 that does not put the INFO tag into the Allow header.  Asterisk 1.4 and all
 versions greater supply the INFO tag as standard.

 Given that 1.2 is in security-only fix mode now, this is not going to be
 changed in SVN or in any subsequent 1.2 release (if any).  You're welcome to
 change the ALLOWED_METHODS define in the top of chan_sip.c and
 recompile, however.

 --
 Tilghman Lesher
 Digium, Inc. | Senior Software Developer
 twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
 Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] Allow Header

[Coco Richard]

Hi all,

In the INVITE from my SIP provider to Asterisk i can see that the
Allow Header includes an INFO Method, but Asterisk replies a 200 OK
with an Allow Header without INFO Method. But in the RFC3261 (20.5)
you can read:

All methods, including ACK and CANCEL, understood by the UA MUST be
included in the list of methods in the Allow header field, when
present. 

My SIP provider seems to refuse to send SIP INFO DTMF and releases the
call, because in 200 OK from * there is no INFO Method in the Allow
Header.

Is that correct.

thx
richard

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Re: [asterisk-users] Allow Header

[Coco Richard]

Hi Alex,

i'm not sure to understand. Asterisk does support SIP INFO, so why
doesn't Asterisk add the INFO Method in the 200OK Response?
richard


On Mon, Nov 9, 2009 at 6:38 PM, Alex Balashov abalas...@evaristesys.com wrote:
 Yes, it's correct.  Asterisk needs to advertise its support of that
 method in order for the other UA to be willing to send messages with
 that request method to it.

 Coco Richard wrote:

 Hi all,

 In the INVITE from my SIP provider to Asterisk i can see that the
 Allow Header includes an INFO Method, but Asterisk replies a 200 OK
 with an Allow Header without INFO Method. But in the RFC3261 (20.5)
 you can read:

 All methods, including ACK and CANCEL, understood by the UA MUST be
 included in the list of methods in the Allow header field, when
 present. 

 My SIP provider seems to refuse to send SIP INFO DTMF and releases the
 call, because in 200 OK from * there is no INFO Method in the Allow
 Header.

 Is that correct.

 thx
 richard

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 Direct  : (+1) (678) 954-0671

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[asterisk-users] RFC 3578 in Asterisk

[Coco Richard]

Hi all,

our asterisk is connected to a sip proxy through a sip trunk. Let's say we
have following dial plan (only an example)

[from_sip_proxy]
exten = 36122512,1,Answer()
exten = 36122512,2,VoiceMailMain()

exten = 3612252,1,Answer()
exten = 3612252,2,MeetMe(313,MI)
exten = 3612252,3,HangUp()

exten = 36122530,1,Answer()
exten = 36122530,2,MusicOnHold()

Overlap from pstn works fine and you can see that asterisk answers with 484
address incomplete as long there is no match.
But if we change our dial plan like the following (we have different
extensions with different length)

[from_sip_proxy]
exten = _36122.,1,Goto(local,${EXTEN:5},1)

[local]
exten = 512,1,Answer()
exten = 512,2,VoiceMailMain()

exten = 52,1,Answer()
exten = 52,2,MeetMe(313,MI)
exten = 52,3,HangUp()

exten = 530,1,Answer()
exten = 530,2,MusicOnHold()

We can notice that incoming calls (e.g for 36122512) are now routed by
asterisk from context [from_sip_proxy] to context [local] and overlap
doesn't work anymore. The answer is 603 Declined.

[CLI]
Sep  4 15:15:21] WARNING[28382]: pbx.c:2450 __ast_pbx_run: Channel
'SIP/192.168.148.186-08c16fe0' sent into invalid extension '5' in context
'local', but no invalid handler
[/CLI]

We think that here the answer for the INVITE 361225 should also be 484
address incomplete and same thing for the next INVITE for 3612251 and finaly
100 Trying for the last INVITE 36122512. Can anyone please confirm this.

thx in advance.
rich
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Re: [asterisk-users] pick up IAX2 calls

[coco]

Hello

I asked the same thing some time ago, but nobody answered.
I founded some workaround.

Use this in your dialplan:
exten = _7.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:1})
exten = _7.,n,Pickup(${EXTEN:[EMAIL PROTECTED])

This worked for me.

Cosmin



--- On Thu, 11/27/08, Bruno Castelo Branco [EMAIL PROTECTED] wrote:
From: Bruno Castelo Branco [EMAIL PROTECTED]
Subject: Re: [asterisk-users] pick up IAX2 calls
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Thursday, November 27, 2008, 4:59 AM




  
Somebody know some work around for it?

I still trying to find a solution but nothing seems to work



thanks



Eric ManxPower Wieling wrote:

  The problem is that IAX2 does not seem to support call pickup.

Bruno Castelo Branco wrote:
  
  
hi
I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 
for all IAX extensions in iax.conf. Didn't works for while.
thanks

Tim Panton wrote:


  I think it doesn't work across channel types.
So it works (if I recall correctly) in IAX or in SIP or in ZAP,
but not in  mixture.

I think that if you have a Dial() that rings several extens,
then any of the technologies involved can pickup with *8

So if you have Dial(IAX/fredSIP/billzap/mark)
then someone in the same group as fred can pickup with IAX
and someone in the same group as bill can pickup with SIP
etc.

So it's an asterisk thing, not an IAX thing per-se.

Tim.

P.S.
(you could try putting in a dummy 'fred' entry into Dial and iax.conf.)
T.

On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote:

  
  
hi

thanks Luis , but doesn't work.
For SIP extensions works well *8, but for IAX a tried *8 and ** + iax 
extension and didn't works

Luis Morales wrote:


  Try with ** + iax extension

Regards,

Luis Morales

On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
 
  
  
Hi

Somebody knows if pickup call works with IAX2?
I enable *8 in features.conf, but doesn't works with IAX2 extensions.
Any idea?

thanks



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Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX

[coco]

 Hello
 
Could you please help me understand if this behavior is corect or not?
I did not find something that says that from iax channels i cannot pickup 
ringing ext using the feature defined in features.conf.
Should I open a bug at Digium?
Any of you tryed this feature and worked? so that i could understand if I am 
doing something wrong,
 
So, if anyody used this feature and worked, please tell me so I can understand,
If not, and is a bug, please place your oppinions.
 
Regards,
Cosmin
 
 
 
Hello Cosmin,

I also tried this, and it doesn't work. I think it is a bug but i'm not sure. 
Let us know if you find any solution.

Regards,
Serghei Gutanu



Cosmin Nistor wrote:






 Hello and thank you for replyes.
 
Eric, I looked for it on the mailing list and google and did not find something 
relevant to be 100% sure that this feature is not supported.
 
Some information clare I founded in 
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups where 
it says that for IAX channels I can use the pickup feature from features.conf.
 
I was looking for an anser to understand if this is supported or not, not to 
lose more time trying to make it work.
 
Shazaum , thank you for your anser, the application Pickup works ok. 
My problem is that this application issued from the dial-plan is
directed pickup, thos means that I have to know the exten that is 
ringing.

I have difficulties because I an using call queues and the channel is not 
anymore only the exten that is ringing, and if I want to pikup a call that is 
comming from a queue, I cannot do this with app Pickup(at least I did not find 
any way to do this--any help from somebody who did is apreciated.)
 
Also, since IAX is developed by asterisk, is strange that for SIP there is 
support, and for IAX, this kind of application is not supported--this is why I 
asked, maybe I am doing something wrong.
 
In this case(if it is not supportted), shoul we/I open a bug repot to Digium? 
 
Botton line, what i am trying to do is to pickup any call that cames in, direct 
call, transfered call, queue call, using IAX, and I am wondering if this is 
possible in any way.
 
Regards,
Cosmin
 
 
I believe chan_iax2 does not support call pickup.  Search the archives.

Shazaum wrote:

 already tested with an exten?
 ex:
 exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
 exten = _*8.,n,Hangup()
 
 2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 
 Hello list
 
  
 
 I am trying to configure a PBX using Asterisk.
 
 The problem I am havong is the following: I want to use the *8 from
 features.conf to pickup any ringing extension from a group, becouse
 I want to put the users in call queues and I want anybody from the
 company to be able to pick a ringing channel, even if is in a queue.
 
  
 
 Whwn using Sip protocol for the users, everithing is going fine, I
 can pickup any ringing extension from the group using *8.
 
 But the problem appears when I am using IAX protocol. When issuing
 *8 from the IAX phone, asterisk tryes to find the *8 in the dialling
 rules returning:
 
  
 
 *CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
 http://10.0.0.30:4569
 [Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
 Rejected connect attempt from 10.0.0.30 http://10.0.0.30, request
 '[EMAIL PROTECTED]' mailto:[EMAIL PROTECTED] does not exist
 
 This I think is wrong, is something like asterisk cannot read from
 features.
 
 With the same setting, when using SIP, i get:
 
  
 
 *CLI   == Using SIP RTP CoS mark 5
 [Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
 handle_request_invite: Nothing to pick up for
 [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
 
 and it works ok.
 
  
 
 I am wondering if any had this problem before and if you can help me
 figure it out(how to make it work--or if is a bug), or find a
 sollution using the app pickup.
 
  
 
 I tryed using asterisk 1.4.13, asterisk 1.4.21.2 http://1.4.21.2,
 asterisk 1.6-rc6 and always the same problem ocurs.
 
  
 
  
 
 Regards,
 
 Cosmin




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Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX

[coco]

 Hello and thank you for replyes.
 
Eric, I looked for it on the mailing list and google and did not find something 
relevant to be 100% sure that this feature is not supported.
 
Some information clare I founded in 
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups where 
it says that for IAX channels I can use the pickup feature from features.conf.
 
I was looking for an anser to understand if this is supported or not, not to 
lose more time trying to make it work.
 
Shazaum , thank you for your anser, the application Pickup works ok. 
My problem is that this application issued from the dial-plan is
directed pickup, thos means that I have to know the exten that is 
ringing.

I have difficulties because I an using call queues and the channel is not 
anymore only the exten that is ringing, and if I want to pikup a call that is 
comming from a queue, I cannot do this with app Pickup(at least I did not find 
any way to do this--any help from somebody who did is apreciated.)
 
Also, since IAX is developed by asterisk, is strange that for SIP there is 
support, and for IAX, this kind of application is not supported--this is why I 
asked, maybe I am doing something wrong.
 
In this case(if it is not supportted), shoul we/I open a bug repot to Digium? 
 
Botton line, what i am trying to do is to pickup any call that cames in, direct 
call, transfered call, queue call, using IAX, and I am wondering if this is 
possible in any way.
 
Regards,
Cosmin
 
 
I believe chan_iax2 does not support call pickup.  Search the archives.

Shazaum wrote:

 already tested with an exten?
 ex:
 exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
 exten = _*8.,n,Hangup()
 
 2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 
 Hello list
 
  
 
 I am trying to configure a PBX using Asterisk.
 
 The problem I am havong is the following: I want to use the *8 from
 features.conf to pickup any ringing extension from a group, becouse
 I want to put the users in call queues and I want anybody from the
 company to be able to pick a ringing channel, even if is in a queue.
 
  
 
 Whwn using Sip protocol for the users, everithing is going fine, I
 can pickup any ringing extension from the group using *8.
 
 But the problem appears when I am using IAX protocol. When issuing
 *8 from the IAX phone, asterisk tryes to find the *8 in the dialling
 rules returning:
 
  
 
 *CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
 http://10.0.0.30:4569
 [Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
 Rejected connect attempt from 10.0.0.30 http://10.0.0.30, request
 '[EMAIL PROTECTED]' mailto:[EMAIL PROTECTED] does not exist
 
 This I think is wrong, is something like asterisk cannot read from
 features.
 
 With the same setting, when using SIP, i get:
 
  
 
 *CLI   == Using SIP RTP CoS mark 5
 [Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
 handle_request_invite: Nothing to pick up for
 [EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED]
 
 and it works ok.
 
  
 
 I am wondering if any had this problem before and if you can help me
 figure it out(how to make it work--or if is a bug), or find a
 sollution using the app pickup.
 
  
 
 I tryed using asterisk 1.4.13, asterisk 1.4.21.2 http://1.4.21.2,
 asterisk 1.6-rc6 and always the same problem ocurs.
 
  
 
  
 
 Regards,
 
 Cosmin
 
 
 
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[asterisk-users] Problem with pickup extension *8 from features.conf using IAX

[coco]

Hello list
 
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from 
features.conf to pickup any ringing extension from a group, becouse I want to 
put the users in call queues and I want anybody from the company to be able to 
pick a ringing channel, even if is in a queue.
 
Whwn using Sip protocol for the users, everithing is going fine, I can pickup 
any ringing extension from the group using *8.
But the problem appears when I am using IAX protocol. When issuing *8 from the 
IAX phone, asterisk tryes to find the *8 in the dialling rules returning:
 
*CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
[Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process: Rejected 
connect attempt from 10.0.0.30, request '[EMAIL PROTECTED]' does not exist

This I think is wrong, is something like asterisk cannot read from features.
With the same setting, when using SIP, i get:
 
*CLI   == Using SIP RTP CoS mark 5
[Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092 handle_request_invite: 
Nothing to pick up for [EMAIL PROTECTED]

and it works ok.
 
I am wondering if any had this problem before and if you can help me figure it 
out(how to make it work--or if is a bug), or find a sollution using the app 
pickup.
 
I tryed using asterisk 1.4.13, asterisk 1.4.21.2, asterisk 1.6-rc6 and always 
the same problem ocurs.
 
 
Regards,
Cosmin


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[asterisk-users] asterisk and 802.1Q

[Coco Richard]

Hi all,

How can i use different VLANs for signaling and audio, e.g vlan 100 for sip
and vlan 200 for rtp? Where can i find documentations for this?

Comments and suggestions are welcomed (a sample config too :-)))

thx in advance
rich
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[asterisk-users] call on hold--hokk flash---i want to know if i can disable it

[coco]

Hello
 
 
 I have a problem with my asterisk server, I want to disable the call on hold 
function when flash hook is pressed.(actually to fully disable it for the users 
connected to the box)
 It does call on hold when I use the asterisk as a rtp proxy, when it does 
nattive bridging, the box has no control over the call and everithing is ok.
 If someone got this problem and solved it, let me know
 
 I am using the users in realtime with mysql, in users.conf i added:(in the 
general section)
 callwaiting = no
 threewaycalling = no
 callwaitingcallerid = no
 transfer = no
 canpark = no
 cancallforward = no
 callreturn = yes
 
 The box is acting the same way
 At list if a call is left in hold, and after the exten hang up, it would ring 
back, it would be something, but it does not, and there is a big chance a call 
gets connected for an unlimited period.
 
 I use asterisk 1.4.13
 
 Do i have a choise if i want to do transcoding(rtp proxy) with the sistem to 
disab le this features?
 
 
 
 Thank you
 
 
 cosmin
 
   
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[asterisk-users] sip tcp support

[richard Coco]

Hi all,

i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the call fails with SIP/2.0 603 Declined.
The strange thing is that the first INVITE uses tcp
and the response is a 100 TRYING, the next 7 INVITE
are using udp and the respose is ICMP Destination
unreachable (Port unreachable).

any hints? Thx in advance

Xtenasterisk HiPath
| INVITE|   |
|--|   |
| TRYING|   |
|--|   |
|   | INVITE|
|   |--|
|   | INVITE|
|   |--|
|   | INVITE|
|   |--|
|   | INVITE|
|   |--|
|   | INVITE|
|   |--|
|   | INVITE|
|   |--|
|   | INVITE|
|   |--|
|603 DECINE |   |
|--|   |
| ACK   |   |
|--|   |


- Registered SIP '971' at 10.4.5.1 port 5060 expires
120
proxy*CLI sip show peer 971

  * Name   : 971
  Secret   : Not set
  MD5Secret: Not set
  Context  : from_hipath
  Subscr.Cont. : Not set
  Language :
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic  : Yes
  Callerid :  
  Expire   : 113
  Insecure : no
  Nat  : RFC3581
  ACL  : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr-IP : 10.4.5.1 Port 5060
  Defaddr-IP  : 0.0.0.0 Port 5060
  Sock fd  : 24
  Transport: TCP
  Def. Username: 971
  SIP Options  : (none)
  Codecs   : 0xc (ulaw|alaw)
  Codec Order  : (alaw,ulaw)
  Status   : Unmonitored
  Useragent: HiPath 4000 V3.0 M5T SIP-UA
SAFE/v3.6.6.10
  Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=tcp

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Re: [asterisk-users] sip tcp support

[richard Coco]

--- J. Oquendo [EMAIL PROTECTED] wrote:

 richard Coco wrote:
  Hi all,
 
  i have asterisk 1.2.17 with sip tcp support and i
 am
  trying to connect asterisk with HiPath 4000 V.3.0
  using SIP. I can see the registration from the
 HG3540.
  But when i try to place a call from Asterisk to
  HiPath, the call fails with SIP/2.0 603 Declined.
  The strange thing is that the first INVITE uses
 tcp
  and the response is a 100 TRYING, the next 7
 INVITE
  are using udp and the respose is ICMP Destination
  unreachable (Port unreachable).
 
  any hints? Thx in advance
 
  Xtenasterisk HiPath
  | INVITE|   |
  |--|   |
  | TRYING|   |
  |--|   |
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |603 DECINE |   |
  |--|   |
  | ACK   |   |
  |--|   |
 
 
  - Registered SIP '971' at 10.4.5.1 port 5060
 expires
  120
  proxy*CLI sip show peer 971
 
* Name   : 971
Secret   : Not set
MD5Secret: Not set
Context  : from_hipath
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
CallingPres  : Presentation Allowed, Not
 Screened
Callgroup:
Pickupgroup  :
Mailbox  :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit   : 0
Dynamic  : Yes
Callerid :  
Expire   : 113
Insecure : no
Nat  : RFC3581
ACL  : No
CanReinvite  : Yes
PromiscRedir : No
User=Phone   : No
Trust RPID   : No
Send RPID: No
DTMFmode : rfc2833
LastMsg  : 0
ToHost   :
Addr-IP : 10.4.5.1 Port 5060
Defaddr-IP  : 0.0.0.0 Port 5060
Sock fd  : 24
Transport: TCP
Def. Username: 971
SIP Options  : (none)
Codecs   : 0xc (ulaw|alaw)
Codec Order  : (alaw,ulaw)
Status   : Unmonitored
Useragent: HiPath 4000 V3.0 M5T SIP-UA
  SAFE/v3.6.6.10
Reg. Contact :
 sip:[EMAIL PROTECTED]:5060;transport=tcp
 
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i have read somewhere that the HG3540 only works with
sip tcp for SIPQ.


http://lists.digium.com/mailman/listinfo/asterisk-users
 

 
 Just out of sheer curiousity, I'm wondering why you
 decided to use TCP 
 as opposed to UDP.
 
 Please don't tell me its for security reasons...
 Just a question.
 
 -- 
 
 J. Oquendo

http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
 sil . infiltrated @ net http://www.infiltrated.net 
 
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Re: [asterisk-users] sip tcp support

[richard Coco]

strange i have:
udp0  0 0.0.0.0:5060   
0.0.0.0:*   9722/asterisk


972 is the tie access code from Hiapth to Asterisk.

--- Tzafrir Cohen [EMAIL PROTECTED] wrote:

 On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard
 Coco wrote:
  
  Hi all,
  
  i have asterisk 1.2.17 with sip tcp support and i
 am
  trying to connect asterisk with HiPath 4000 V.3.0
  using SIP. I can see the registration from the
 HG3540.
  But when i try to place a call from Asterisk to
  HiPath, the call fails with SIP/2.0 603 Declined.
  The strange thing is that the first INVITE uses
 tcp
  and the response is a 100 TRYING, the next 7
 INVITE
  are using udp and the respose is ICMP Destination
  unreachable (Port unreachable).
 
 Anybody listens on the UDP port?
 
   netstat -lnup | grep 5060
 
  
  any hints? Thx in advance
  
  Xtenasterisk HiPath
  | INVITE|   |
  |--|   |
  | TRYING|   |
  |--|   |
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |   | INVITE|
  |   |--|
  |603 DECINE |   |
  |--|   |
  | ACK   |   |
  |--|   |
  
  
  - Registered SIP '971' at 10.4.5.1 port 5060
 expires
  120
 
 Should that message be changed to reflect the fact
 that the port is TCP?
 (and is it for a TCP port indeed?)
 
  proxy*CLI sip show peer 971
  
* Name   : 971
Secret   : Not set
MD5Secret: Not set
Context  : from_hipath
Subscr.Cont. : Not set
Language :
AMA flags: Unknown
CallingPres  : Presentation Allowed, Not
 Screened
Callgroup:
Pickupgroup  :
Mailbox  :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit   : 0
Dynamic  : Yes
Callerid :  
Expire   : 113
Insecure : no
Nat  : RFC3581
ACL  : No
CanReinvite  : Yes
PromiscRedir : No
User=Phone   : No
Trust RPID   : No
Send RPID: No
DTMFmode : rfc2833
LastMsg  : 0
ToHost   :
Addr-IP : 10.4.5.1 Port 5060
Defaddr-IP  : 0.0.0.0 Port 5060
Sock fd  : 24
Transport: TCP
Def. Username: 971
SIP Options  : (none)
Codecs   : 0xc (ulaw|alaw)
Codec Order  : (alaw,ulaw)
Status   : Unmonitored
Useragent: HiPath 4000 V3.0 M5T SIP-UA
  SAFE/v3.6.6.10
Reg. Contact :
 sip:[EMAIL PROTECTED]:5060;transport=tcp
 
 -- 
Tzafrir Cohen   
 icq#16849755   
 jabber:[EMAIL PROTECTED]
 +972-50-7952406  
 mailto:[EMAIL PROTECTED]   
 http://www.xorcom.com 
 iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] sip tcp support

[richard Coco]

sorry, it works with upd... I am now able to make and
to receive calls.

thx...

--- richard Coco [EMAIL PROTECTED] wrote:

 
 
 strange i have:
 udp0  0 0.0.0.0:5060   
 0.0.0.0:*  
 9722/asterisk
 
 
 972 is the tie access code from Hiapth to Asterisk.
 
 --- Tzafrir Cohen [EMAIL PROTECTED] wrote:
 
  On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard
  Coco wrote:
   
   Hi all,
   
   i have asterisk 1.2.17 with sip tcp support and
 i
  am
   trying to connect asterisk with HiPath 4000
 V.3.0
   using SIP. I can see the registration from the
  HG3540.
   But when i try to place a call from Asterisk to
   HiPath, the call fails with SIP/2.0 603
 Declined.
   The strange thing is that the first INVITE uses
  tcp
   and the response is a 100 TRYING, the next 7
  INVITE
   are using udp and the respose is ICMP
 Destination
   unreachable (Port unreachable).
  
  Anybody listens on the UDP port?
  
netstat -lnup | grep 5060
  
   
   any hints? Thx in advance
   
   Xten  asterisk HiPath
   | INVITE|   |
   |--| |
   | TRYING|   |
   |--| |
   |   | INVITE|
   |   |--|
   |   | INVITE|
   |   |--|
   |   | INVITE|
   |   |--|
   |   | INVITE|
   |   |--|
   |   | INVITE|
   |   |--|
   |   | INVITE|
   |   |--|
   |   | INVITE|
   |   |--|
   |603 DECINE |   |
   |--| |
   | ACK   |   |
   |--| |
   
   
   - Registered SIP '971' at 10.4.5.1 port 5060
  expires
   120
  
  Should that message be changed to reflect the fact
  that the port is TCP?
  (and is it for a TCP port indeed?)
  
   proxy*CLI sip show peer 971
   
 * Name   : 971
 Secret   : Not set
 MD5Secret: Not set
 Context  : from_hipath
 Subscr.Cont. : Not set
 Language :
 AMA flags: Unknown
 CallingPres  : Presentation Allowed, Not
  Screened
 Callgroup:
 Pickupgroup  :
 Mailbox  :
 VM Extension : asterisk
 LastMsgsSent : 32767/65535
 Call limit   : 0
 Dynamic  : Yes
 Callerid :  
 Expire   : 113
 Insecure : no
 Nat  : RFC3581
 ACL  : No
 CanReinvite  : Yes
 PromiscRedir : No
 User=Phone   : No
 Trust RPID   : No
 Send RPID: No
 DTMFmode : rfc2833
 LastMsg  : 0
 ToHost   :
 Addr-IP : 10.4.5.1 Port 5060
 Defaddr-IP  : 0.0.0.0 Port 5060
 Sock fd  : 24
 Transport: TCP
 Def. Username: 971
 SIP Options  : (none)
 Codecs   : 0xc (ulaw|alaw)
 Codec Order  : (alaw,ulaw)
 Status   : Unmonitored
 Useragent: HiPath 4000 V3.0 M5T SIP-UA
   SAFE/v3.6.6.10
 Reg. Contact :
  sip:[EMAIL PROTECTED]:5060;transport=tcp
  
  -- 
 Tzafrir Cohen   
  icq#16849755   
  jabber:[EMAIL PROTECTED]
  +972-50-7952406  
  mailto:[EMAIL PROTECTED]   
  http://www.xorcom.com 
  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] install and setup app_mp4 application

[richard Coco]

Hi all,

according to
http://sip.fontventa.com/content/view/15/44/ i have
compiled the mpeg4ip libries without problem. After
copying the app_mp4.c file into de Asterisk apps
directory and changing the Makefile like.

[...]
app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $
${CYGSOLIB} -lodbc

app_mp4.so : app_mp4.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $
${CYGSOLIB} -lmp4 -lmp4v2

ifeq (SunOS,$(shell uname))
app_chanspy.so: app_chanspy.o
$(CC) $(SOLINK) -o $@ $ -lrt
endif
[...]

i get following error.
Mar 21 19:08:22 WARNING[26686]: pbx.c:1720
pbx_extension_helper: No application 'mp4save' for
extension...

it seems that after the recompilation of asterisk no
app_mp4.o/app_mp4.so is created in ../asterisk/apps/.

asterisk# ls apps/app_mp*
apps/app_mp3.c  apps/app_mp3.o  apps/app_mp3.so 
apps/app_mp4.c

has anyone an idea...
thx in advabce...
  


 

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Re: [asterisk-users] voicemail scenario

[richard Coco]

Hi,

some additional informations what i am trying to do.
In the voicemail.conf you have several setting for the
[general] section. One is the emailsubject. I have
something like emailsubject=New voicemail for
${VM_NAME}. In my [contexts] i have.

[context_section] 
extension_number =
voicemail_password,user_name,user_email_address,user_pager_email_address,user_option(s)

  
The user_option(s) field can be used to override
default settings defined in the general section. There
are nine settings which may be used... unfortunately
not emailsubject. My question is, is there an
alternative way to override the default setting for
emailsubject defined in the [general]

thx.

--- Dovid B [EMAIL PROTECTED] wrote:

 I dont think you can but you can use a variable.
 Have a look at 
 voicemail.conf. You can edit the message the
 asterisk sends out. If you want 
 the CID to be in the subject you can use the
 variable ${CALLERID(number)} .
 
 - Original Message - 
 From: richard Coco [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, March 13, 2007 10:53 PM
 Subject: Re: [asterisk-users] voicemail scenario
 
 
 
 Hi,
 
 i finally managed to get it work using GlobalVar.
 I still have a question. I have several context in
 my
 voicemail.conf like
 [default]
 [customer_1]
 [customer_2]
 [customer_3]
 
 How can i set a different emailsubject for each
 context?
 
 thx
 
 
 --- richard Coco [EMAIL PROTECTED] wrote:
 
  Hi all,
 
  i need help to implement a voicemail scenario.
 What
  i
  am trying to do is the following.
 
  user X dials a direct access for user Y voicemail
  and
  is asked to enter a number (e.g 12345678) and then
  leaves a message. Then asterisk sends a
 notification
  with attachement. The problem is that i need the
  number entered (e.g 12345678) in the subject. Is
  that
  possible.
 
  thx in advance.
 
 
 
 


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[asterisk-users] voicemail scenario

[richard Coco]

Hi all,

i need help to implement a voicemail scenario. What i
am trying to do is the following.

user X dials a direct access for user Y voicemail and
is asked to enter a number (e.g 12345678) and then
leaves a message. Then asterisk sends a notification
with attachement. The problem is that i need the
number entered (e.g 12345678) in the subject. Is that
possible.

thx in advance.


 

Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html 
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Re: [asterisk-users] voicemail scenario

[richard Coco]

Hi,

i finally managed to get it work using GlobalVar. 
I still have a question. I have several context in my
voicemail.conf like
[default]
[customer_1]
[customer_2]
[customer_3]

How can i set a different emailsubject for each
context?

thx


--- richard Coco [EMAIL PROTECTED] wrote:

 Hi all,
 
 i need help to implement a voicemail scenario. What
 i
 am trying to do is the following.
 
 user X dials a direct access for user Y voicemail
 and
 is asked to enter a number (e.g 12345678) and then
 leaves a message. Then asterisk sends a notification
 with attachement. The problem is that i need the
 number entered (e.g 12345678) in the subject. Is
 that
 possible.
 
 thx in advance.
 
 
  


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[asterisk-users] one way audio when forwarding from ser to asterisk

[richard Coco]

Hi all,

i have ser and asterisk on the same box with a public
ip address. When an UA behind NAT registred on SER try
to call the Voicemail or another UA registred on
Asterisk i have one way audio (caller cannot hear the
callee).

[UA/SER]--[router/nat]--[SER/Asterisk]


UA has private IP(192.168.204.19) and public IP is
89.106.xxx.yyy
SER/ASterisk has public ip (89.106.yyy.zzz).

In the sip trace one can see that signaling is ok but
Asterisk sends RTP from 89.106.xxx.zzz to
192.168.204.19 not to 89.106.xxx.yyy

ps: when UA registred on SER try to call UA2 registred
on SER every thing works fine.

how can i fix this issue.
thx   


 

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Re: [asterisk-users] 802.1x support in wired sip hardphones ?

[richard Coco]

 How (and where) could you provision those phones ?
 Do you have any support from Siemens or anyone ?

We have a HiPath4000 V1.0 interconnected with Asterisk
using oh323. I have flashed several OptiPoints (from
the HiPath) to SIP firmware. But again OptiPoints seem
to work well with Asterisk but never use them with
802.1x.

rich.


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Re: [asterisk-users] 802.1x support in wired sip hardphones ?

[richard Coco]

Hi,

http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm

rich.

--- Olivier [EMAIL PROTECTED] wrote:

 Hi,
 
 Is anyone aware of a wired sip hardphone supporting
 802.1x authentication ?
 I've been told some Avaya and Alcatel ip phones
 supported 802.1x.
 
 As 802.1x is widely used with wireless hardphones,
 I'm wondering whether or
 not, 802.1x could also be valuable for wired
 environments.
 
 Regards
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Re: [asterisk-users] 802.1x support in wired sip hardphones ?

[richard Coco]

--- Olivier [EMAIL PROTECTED] wrote:

 Thanks !!
 I've never heard of this one (I mean : I've never
 heard of OptiPoint phones
 to support 802.1x).
 
 Have you used the SIP version with Asterisk and
 802.1x ?
we have several Optipoint410/420/600 configured with
Asterisk and they seem to work well (but no 802.1x).We
made several tests with MacAuthentication last year.

 Am I correct to think that using 802.1x isn't
 directly of Asterisk concern ?

802.1x has nothing to do with Asterisk. You need a
supplicant (your phone) an Authenticator (your switch)
and a authentication server (e.g FreeRadius)

a howto about 802.1X Port-Based Authentication are
avalaible at
http://tldp.org/HOWTO/html_single/8021X-HOWTO/


 
 2007/1/2, richard Coco [EMAIL PROTECTED]:
 
  ***
  This message was sent to your KasMail disposable
 email address:
  Asterisk Users Mailing List - Non-Commercial
 Discussion
  asterisk-users@lists.digium.com
  ***
 
 
  Hi,
 
 
 

http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm
 
  rich.
 
  --- Olivier [EMAIL PROTECTED] wrote:
 
   Hi,
  
   Is anyone aware of a wired sip hardphone
 supporting
   802.1x authentication ?
   I've been told some Avaya and Alcatel ip phones
   supported 802.1x.
  
   As 802.1x is widely used with wireless
 hardphones,
   I'm wondering whether or
   not, 802.1x could also be valuable for wired
   environments.
  
   Regards
   
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Re: [asterisk-users] [Asterisk-Java] SipShowPeerAction

[richard Coco]

Hi again,

i am still missing something 'cause i am not able to
handle the PeerEntryEvent. The other Events are ok.
Here is what i did.

public void run() throws
IOException,
AuthenticationFailedException,
TimeoutException
{
managerConnection.login();
managerConnection.addEventListener(this);

SipShowPeerAction sipShowPeerAction = new
SipShowPeerAction(2001);
managerConnection.sendAction(sipShowPeerAction);

}
public void onManagerEvent(ManagerEvent event)
{
HashMapString, JButton hmap = new HashMapString,
JButton();
hmap.put(SIP/2000, PresenceGUI.sButton2000);
hmap.put(SIP/2001, PresenceGUI.sButton2001);
hmap.put(SIP/2002, PresenceGUI.sButton2002);
hmap.put(SIP/2003, PresenceGUI.sButton2003);
hmap.put(SIP/2004, PresenceGUI.sButton2004);

if (event instanceof PeerEntryEvent)
{

System.out.println(((PeerEntryEvent)event).getStatus());
}
if (event instanceof PeerStatusEvent)
{
if (((PeerStatusEvent)
event).getPeerStatus().equals(PeerStatusEvent.STATUS_REGISTERED))
{

hmap.get(((PeerStatusEvent)event).getPeer()).setIcon(new
ImageIcon(personal_green.png));
}

if (((PeerStatusEvent)
event).getPeerStatus().equals(PeerStatusEvent.STATUS_UNREGISTERED))
{

hmap.get(((PeerStatusEvent)event).getPeer()).setIcon(new
ImageIcon(personal_gray.png));
}
}
if (event instanceof NewChannelEvent)
{
if (((NewChannelEvent)
event).getState().equals(Ringing))
{

hmap.get(((NewChannelEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_red.png));
}
if (((NewChannelEvent)
event).getState().equals(Ring))
{

hmap.get(((NewChannelEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_red.png));
}
}
if (event instanceof HangupEvent)
{

if(((HangupEvent)event).getChannel().substring(0,
5).equals(SIP/2))
{

hmap.get(((HangupEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_green.png));
}
}
}
}


thx in advance!





--- Tim Panton [EMAIL PROTECTED] wrote:

 
 On 4 Oct 2006, at 16:33, richard Coco wrote:
 
  Hi all,
 
  first of all sorry for the question. I know there
 is
  an asterisk-java mailinglist but i am not
 subscribed
  to this list and i am sure there are asterisk-java
  guru on this list who can help me.
 
  I am trying to get the status of a peer using
  SipShowPeerAction. Unfortunately the getStatus
  method gives me everytime null.
 
  SipShowPeerAction sipShowPeerAction = new
  SipShowPeerAction(2001);
  managerConnection.sendAction(sipShowPeerAction);
  PeerEntryEvent peerEntryEvent = new
  PeerEntryEvent(sipShowPeerAction);
  System.out.println(peerEntryEvent.getStatus());
 
  What wrong with this example? Maybe someone can
 give
  me a working example.
 
 The way Java normally works is that you add register
 yourself as an
 event listener, and the framework then sends you an
 event when  
 something happens.
 
 so your class needs to implement
 ManagerEventListener
 then you say something like :
 
 void doit(){
   managerConnection.addEventListener(this)
   SipShowPeerAction sipShowPeerAction =
 newSipShowPeerAction(2001);
   managerConnection.sendAction(sipShowPeerAction);
 
 }
 
 public void onManagerEvent(ManagerEvent event) {
   if (event instanceof PeerEntryEvent){
   

System.out.println(((PeerEntryEvent)event).getStatus());
   } else {
   System.out.println(Some other event);
   }
 }
 
 
 Tim Panton
 
 www.mexuar.com
 
 
 
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[asterisk-users] [Asterisk-Java] SipShowPeerAction

[richard Coco]

Hi all,

first of all sorry for the question. I know there is
an asterisk-java mailinglist but i am not subscribed
to this list and i am sure there are asterisk-java
guru on this list who can help me.

I am trying to get the status of a peer using
SipShowPeerAction. Unfortunately the getStatus
method gives me everytime null.

SipShowPeerAction sipShowPeerAction = new
SipShowPeerAction(2001);
managerConnection.sendAction(sipShowPeerAction);
PeerEntryEvent peerEntryEvent = new
PeerEntryEvent(sipShowPeerAction);
System.out.println(peerEntryEvent.getStatus());

What wrong with this example? Maybe someone can give
me a working example.

hope someone can help...

thx in advance 

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[asterisk-users] unable to change the emailbody for email notification

[richard Coco]

Hi all,

the default message for email notification looks like:

New 0:09 long msg in box 2001
from XliteUser2002, on Monday, September 18, 2006 at
04:24:11 PM

i try to change it with emailbody= but i always get
the default message body.

my voicemail.conf looks like
[general]

format=wav49|gsm|wav
attach=no
maxmessage=180
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3

emailbody=Dear ${VM_NAME}:\n\n\t you were just left a
${VM_DUR} long message (number ${VM_MSGNUM})\nin
mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on
${VM_DATE} \n

[default]
2001 = 2001,2001,Coco Richard,[EMAIL PROTECTED]

Is there something wrong with my config?
thx in advance

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[asterisk-users] IAX phone recommandation

[richard Coco]

Hi all,

we plan to install several IAX softphones.

http://www.voip-info.org/wiki-Asterisk+IAX+clients
lists a lot of IAX phones for Windows and Linux. Which
one would you recommand? We will install IAX client on
Linux and Windows.

thx richard

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Re: [asterisk-users] SIP client with video???

[richard Coco]

Hi,

i have xlite too and it works without any problems.

ps: what about ekiga? (www.ekiga.org)

rich

--- Joao Pereira [EMAIL PROTECTED] wrote:

 Hello to all
 can someone recommend me a nice SIP client with
 video for windows??
 
 I tried X-Lite 3.0 but it's a lousy piece of
 software.
 
 Does someone knows about a better software?
 Regards
 Joao Pereira
 
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[asterisk-users] [oh323]FastStart/H245Tunnelling/H245inSetup

[richard Coco]

Hi all,

i have following setup

[]--[asterisk]--[oh323]--[HiPath]--[8000]


 is my voicemail access
exten = ,1,Answer()
exten = ,2,VoiceMailMain() 

8000 is an Optiset phone registered on the HiPath.
When 8000 calls  i have no voice (depends on the
setting of FastStart). When FastStart=yes in oh323 the
caller can't hear the voivemail message (otherwise
(when FastStart=no) every thing works fine.

Can anyone explain the impact of FastStart?
What is the H245inSetup parameter?

thx in advance...



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Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000

[richard Coco]

Hi again,

the TR6T parameter (i have german settings for my AMO
so it is TR6Q ;-)) resolved the same issue for my...
the difference is that i have an IP-trunk (using
oh323) between Asterisk and the HiPath. Have you tried
to remove the TR6T parameters...

Can you also paste the following outputs from the H4K

DISPLAY-APS:TYPE=PSGL,SP=y0*;
REGENERATE-TDCSU:PEN1=XX-XX-XX-XX; (change xx-xx-xx-xx
to the pin of the isdn trunk e.g 01-02-25-00) 
DISPLAY-COT:COTNO=XX; (change XX to the cot number of
the trunk)

if the log is not to huge please paste the last 30 min
of the history file. Try to reproduce the issue after
that type:

START-HISTA:RTYPE=SEARCHB,STIME=2006-06-27/09:00,ETIME=2006-06-27/09:30;

adjust the start time and the end time in a way that
the test is in the range between STIME and ETIME...

regards rich...

--- Josué Conti [EMAIL PROTECTED] wrote:

 Hi Richard.
 Thank you very much for its attention. In the
 reality what is occurring is
 that in some originated calls of the HiPath with
 destination to the Asterisk
 they are being without the dumb and rings. I do not
 have this parameter in
 my HiPath 4000, what I have seemed in the COT is
 TR6T (1tr6 isdn tie link)
 would be this parameter?
  Best Regards
 Josué
 
 2006/6/26, richard Coco [EMAIL PROTECTED]:
 
 
  Hi Josué
 
  if the Siemens phone calls Asterisk, it didn't get
 a
  dial tone from Asterisk? Is it correct?
 
  if yes, this is depending of Asterisk which didn't
  generates a ringback messages as it expexts dial
 ton
  generation localy. So try this workaround for
 HiPath
  local dial ton generation:
  - Add option TR6Q(TRGT) to the class of trunk
 (COT)
  parameters
 
  hope it will help...
 
  rich
 
 
 
 
 
  --- Josué Conti [EMAIL PROTECTED] wrote:
 
 Hello all.
I have installed and functioning
 asterisk-1.2.9.1
   where I effected one
   upgrade in asterisk-1.0.9, is interconnected
 with a
   PABX Siemens HiPath 4000
   in ISDN PRI with protocol QSIG, the one that is
   happening he is that the
   calls originated for PABX Siemens and destined
 to
   SIP phones asterisk are
   being without audio, nor Ring, is dumb. They
 could
   help in this case me?
   Best Regards
  
   Josué
   
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Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000

[richard Coco]

Hi again...

normally the 0/16 is a d-channel.

check the config in the zapata.conf. You should have
some thing like this

/etc/zapata.conf
bchan=1-15 
dchan=16 
bchan=17-31 

/etc/asterisk/zapata.conf
channel = 1-15,17-31 

i don't rember exactelly but in /proc/zaptel there is
the possibility to check if the channels are in use or
not. Maybe someone else can give you a hint...

sorry but i only interconnect Asterisk and H4K using
chan_capi and i have no experience with zapata ;-(

rich

--- Josué Conti [EMAIL PROTECTED] wrote:

 Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386
 pri_dchannel: Ring requested
 on unconfigured channel 0/16 span 1
 I noticed this message in the CLI, when I tried to
 effect one call of HiPath
 4000 for asterisk. Ring occurred, however when the
 voicemail of asterisk
 took care of call it was dumb, without no sound. I
 thank the attention
 Regards
 
 Josué
 
 2006/6/26, Josué Conti [EMAIL PROTECTED]:
 
   Hi Richard.
  Thank you very much for its attention. In the
 reality what is occurring is
  that in some originated calls of the HiPath with
 destination to the Asterisk
  they are being without the dumb and rings. I do
 not have this parameter in
  my HiPath 4000, what I have seemed in the COT is
 TR6T (1tr6 isdn tie link)
  would be this parameter?  
Best Regards
  Josué
 
  2006/6/26, richard Coco [EMAIL PROTECTED]:
 
  
   Hi Josué
  
   if the Siemens phone calls Asterisk, it didn't
 get a
   dial tone from Asterisk? Is it correct?
  
   if yes, this is depending of Asterisk which
 didn't
   generates a ringback messages as it expexts dial
 ton
   generation localy. So try this workaround for
 HiPath
   local dial ton generation:
   - Add option TR6Q(TRGT) to the class of trunk
 (COT)
   parameters
  
   hope it will help...
  
   rich
  
  
  
  
  
   --- Josué Conti [EMAIL PROTECTED] wrote:
  
  Hello all.
 I have installed and functioning
 asterisk-1.2.9.1
where I effected one
upgrade in asterisk-1.0.9, is interconnected
 with a
PABX Siemens HiPath 4000
in ISDN PRI with protocol QSIG, the one that
 is
happening he is that the
calls originated for PABX Siemens and destined
 to
SIP phones asterisk are
being without audio, nor Ring, is dumb. They
 could
help in this case me?
Best Regards
   
Josué

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Re: [Asterisk-Users] siemens pbx and asterisk

[richard Coco]

Hi,

which Hicom and which version is installed?

Hicom 300 or Hicom100?

rich

--- Lito Lampitoc [EMAIL PROTECTED] wrote:

 Hello all,
 
 I'm new to asterisk. Our company wants to setup an
 asterisk server and will
 eventually move to IP centric phones, but they don't
 want to just throw away
 the old Siemens PBX, so during the process we want
 to integrate it with
 asterisk. Is it possible? and how?
 
 thanks.
 
 Lito
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RE: [Asterisk-Users] Re: siemens pbx and asterisk

[richard Coco]

hi all,

The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.
I'am not sure but i thing that the feature CallerID
Name was introduced in version 3 of the H.323
standard. More informations about the owerviews at
http://www.packetizer.com/voip/h323/.

-Concerning HiPathv3.0.
In version 3.0 the HiPath has a new board (the HG3540)
which supports SIP (for Endpoints) and SIPQ for
SIP-trunking. You are now able to interconnect
Asterisk and HiPath using H.323, ISDN and/or SIPQ.

rich

--- Herchi Silviu [EMAIL PROTECTED] wrote:

 Hi,
 
 As I wrote, the HiPath needs to be upgraded to
 version 3 (don't ask me any details, I'm not a
 Siemens expert) in order to have the CallerID name
 passed over the H.323 link. Earlier versions (my
 case) ony sends and accepts the CallerId number.
 
 I have set up a workaround for calls coming to
 Asterisk: an AGI script sets the CallerID name
 according to their CallerID number by looking it up
 in a database. This is done in real time for every
 incoming call. Obviously it doesn't work for calls
 going from Asterisk to the HiPath.
 
 Regards,
 
 Silviu
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On
 Behalf Of Michael Hamann
 Sent: 27 June 2006 14:58
 To: Asterisk Users Mailing List - Non-Commercial
 Discussion
 Cc: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Re: siemens pbx and
 asterisk
 
 Hi Silviu,
 
 did you manage to get the callername to work? I have
 a comparable setup with a hipath System but I
can�t
 get the callername to be displayed over the trunk.
 The callernumber works but not the name...
 
 Any suggestion?
 
 Thanks
 Michael
 
 
  We have successfully integrated an existing
 Siemens HiPath 4500 PBX 
  with two Asterisk servers.
 
  On the first one we use a H.323 trunk (it needs a
 card on the PBX, I 
  think it's called HG3550). It works pretty well,
 except for one 
  missing feature - the callerid name is not
 transmitted over the link 
  (it is a limitation of the PBX that should
 disappear when it is 
  upgraded to the
  V3 version). The nice thing is it doesn't take any
 special hardware on 
  the Asterisk server - you just have to compile and
 setup an H.323 
  channel (asterisk-oh323 works best for us).
 
  On the second one we have a Digium TE110P
 connected to the PBX using a 
  PRI. It works well too, you just need the PBX to
 have a trunk defined 
  and you're ready to go. We only use ten channels,
 so I can't say if 
  the performance is better. In this case you need
 libpri and zaptel on 
  the Asterisk.
 
  I hope this helps,
 
  Silviu
 
 
  ---
  Hello all,
 
  I'm new to asterisk. Our company wants to setup an
 asterisk server and 
  will eventually move to IP centric phones, but
 they don't want to just 
  throw away the old Siemens PBX, so during the
 process we want to 
  integrate it with asterisk. Is it possible? and
 how?
  thanks.
  Lito
 
 
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[Asterisk-Users] Re: Asterisk-1.2.9.1 with Siemens HiPath 4000

[richard Coco]

Hi Josue...

i have taken a short look at the configuration you
sent to me off list. 

First of all, try to change the protocol from ECMAV2
to ETSI or EDSS1 (set the segmentation to 1) and like
suggested by Silviu change the switchtype=EuroISDN
too. EcmaV2 is normaly used to interconnect Siemens
PBXs.

Something strange is that in the HISTA you have
severals CIRCUIT  EXT DIALTONE ERROR from the TM2LP
(analog trunk line).

i will compare tomorrow the COT with the one we have
configured at the office...

rich

--- Josue Conti [EMAIL PROTECTED] wrote:

 Hi Richard
 Thank you very much for its attention
 Below my configurations of HiPath the 4000 and
 asterisk-1.2.9.1 Agrade�o its
 attention, if to need something to communicate
 itself


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Re: [Asterisk-Users] Asterisk-1.2.9.1 with Siemens HiPath 4000

[richard Coco]

Hi Josué

if the Siemens phone calls Asterisk, it didn't get a
dial tone from Asterisk? Is it correct?

if yes, this is depending of Asterisk which didn't
generates a ringback messages as it expexts dial ton
generation localy. So try this workaround for HiPath
local dial ton generation:
- Add option TR6Q(TRGT) to the class of trunk (COT)
parameters

hope it will help...

rich





--- Josué Conti [EMAIL PROTECTED] wrote:

   Hello all.
  I have installed and functioning asterisk-1.2.9.1
 where I effected one
 upgrade in asterisk-1.0.9, is interconnected with a
 PABX Siemens HiPath 4000
 in ISDN PRI with protocol QSIG, the one that is
 happening he is that the
 calls originated for PABX Siemens and destined to
 SIP phones asterisk are
 being without audio, nor Ring, is dumb. They could
 help in this case me?
 Best Regards
 
 Josué
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Re: [Asterisk-Users] SIP voice recorder

[richard Coco]

Hi,

maybe http://www.oreka.org


--- Vic [EMAIL PROTECTED] wrote:

 Hi, I was wondering if anyone knows of a opensource
 SIP
 voice logger. 
 
 I need to simultaneously record around 150 to 200
 sessions.
 
 I figured that if I just set a mirroring port on the
 switch and just send all RTP packets to it, I would
 be
 able to do it.  The problem is: has anyone done it
 before?
 Is there a better way to do it?
 
 Thanks in advance!
 Vic
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Re: [Asterisk-Users] SIP voice recorder

[richard Coco]

hi,

maybe http://www.oreka.org


--- Vic [EMAIL PROTECTED] wrote:

 Hi, I was wondering if anyone knows of a opensource
 SIP
 voice logger. 
 
 I need to simultaneously record around 150 to 200
 sessions.
 
 I figured that if I just set a mirroring port on the
 switch and just send all RTP packets to it, I would
 be
 able to do it.  The problem is: has anyone done it
 before?
 Is there a better way to do it?
 
 Thanks in advance!
 Vic
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[Asterisk-Users] no SUBSCRIBE request sent

[richard Coco]

Hi all,

i am playing around with several optipoint4x0 and run
into trouble trying to get hint functionality to work.

I notice that there is no status notifications. But
afaik this should be implemented via the
SUBSCRIBE/NOTIFY mechanism.

I can see INVITE, TRYING, RINGING, ACK, BYE but no
SUBSBCRIBE in my sip debug traces.

I have problem to understand how hint priority works.
I follow the instructions from
http://www.voip-info.org/wiki/index.php
page=Asterisk+standard+extensions but it still doesn't
work.

[2001]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=notify
disallow=all
allow=alaw
allow=ulaw

[2002]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=notify
disallow=all
allow=alaw
allow=ulaw

[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2002,1,Dial(SIP/2002,10,tr)

[notify]
exten = 2001,hint,SIP/2001
exten = 2002,hint,SIP/2002


thx in advance for your help.



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Re: [Asterisk-Users] no SUBSCRIBE request sent

[richard Coco]

Hi,

first of all, sorry for this long thread... I have
changed my extensions.conf like you suggested and
delete the line with subscribecontext=notify. But
unfortunately i still don't see subscribe request in
the sip debug trace.

SIP Debugging enabled
kingcoco*CLI
-- SIP read from 192.168.204.5:6108:


--- (0 headers 0 lines) Nat keepalive ---
kingcoco*CLI
-- SIP read from 192.168.204.100:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 70
Content-Length: 307
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK4c6bfc983
Call-ID: 7e6c264483fd010
From: OptiPoint410std
sip:[EMAIL PROTECTED];tag=c2a05e95916bbfa;epid=SC22390b
To: sip:[EMAIL PROTECTED]
CSeq: 1 INVITE
Supported: timer
Min-SE: 90
Supported: 100rel
Allow-Events: talk, hold, conference
Allow:
INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE
Content-Type: application/sdp
Contact: OptiPoint410std
sip:[EMAIL PROTECTED]:5060;transport=udp
Supported: replaces
User-Agent: optiPoint 410_420/v4 4.1.66

v=0
o=MxSIP 0 1595508908 IN IP4 192.168.204.100
s=SIP Call
c=IN IP4 192.168.204.100
t=0 0
m=audio 5004 RTP/AVP 9 8 0 18 4 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (17 headers 14 lines)---
Using INVITE request as basis request -
7e6c264483fd010
Sending to 192.168.204.100 : 5060 (non-NAT)
Found user '2001'
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 192.168.204.100:5004
Found description format G722
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined -
0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event),
peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
Looking for 2002 in local (domain 192.168.204.223)
list_route: hop:
sip:[EMAIL PROTECTED]:5060;transport=udp
Transmitting (no NAT) to 192.168.204.100:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100
From: OptiPoint410std
sip:[EMAIL PROTECTED];tag=c2a05e95916bbfa;epid=SC22390b
To: sip:[EMAIL PROTECTED]
Call-ID: 7e6c264483fd010
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


---
-- Executing Dial(SIP/2001-65fe,
SIP/2002|10|tr) in new stack
We're at 192.168.204.223 port 10830
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 192.168.204.5:6108:
INVITE sip:[EMAIL PROTECTED]:6108 SIP/2.0
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK153d90a3;rport
From: OptiPoint410std
sip:[EMAIL PROTECTED];tag=as29a3f9ee
To: sip:[EMAIL PROTECTED]:6108
Contact: sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 17 May 2006 08:58:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 246

v=0
o=root 24071 24071 IN IP4 192.168.204.223
s=session
c=IN IP4 192.168.204.223
t=0 0
m=audio 10830 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 2002
Transmitting (no NAT) to 192.168.204.100:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100
From: OptiPoint410std
sip:[EMAIL PROTECTED];tag=c2a05e95916bbfa;epid=SC22390b
To: sip:[EMAIL PROTECTED];tag=as5094780f
Call-ID: 7e6c264483fd010
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


---
kingcoco*CLI
-- SIP read from 192.168.204.5:6108:
SIP/2.0 180 Ringing
To: sip:[EMAIL PROTECTED]:6108;tag=0a630b27
From:
OptiPoint410stdsip:[EMAIL PROTECTED];tag=as29a3f9ee
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: sip:[EMAIL PROTECTED]:6108
Content-Length: 0


--- (8 headers 0 lines)---
-- SIP/2002-7bc1 is ringing
kingcoco*CLI
-- SIP read from 192.168.204.5:6108:
SIP/2.0 200 OK
To: sip:[EMAIL PROTECTED]:6108;tag=0a630b27
From:
OptiPoint410stdsip:[EMAIL PROTECTED];tag=as29a3f9ee
Via: SIP/2.0/UDP
192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: sip:[EMAIL PROTECTED]:6108
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 185

v=0
o=- 10603328 

Re: [Asterisk-Users] no SUBSCRIBE request sent

[richard Coco]

Hi again,

what do you mean exactely with Have you configured
your phone to subscribe to the extension? :).

I have several optipoint410 and eyebeam. On one of the
Optipoint(exten 2001) i have configured a selected
dialing bottum with the extensions of the
eyebeam(exten 2002). Do i need more configuration on
the IP-phone?

thx in advance

--- Avi Miller [EMAIL PROTECTED] wrote:

 
 On 17/05/2006, at 8:27 PM, richard Coco wrote:
 
  unfortunately i still don't see subscribe request
 in
  the sip debug trace.
 
 Have you configured your phone to subscribe to the
 extension? :)
 
 cYa,
 Avi
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Re: [Asterisk-Users] Hint priority

[richard Coco]

Hi,

i have change my sip.conf and my extensions.conf but
unfortunately nothing change. Should i not see the
hint priority in the CLI?

richard

--- Steve Davies [EMAIL PROTECTED] wrote:

 On 5/12/06, Jerry Jones [EMAIL PROTECTED] wrote:
  I believe the hint priority must be in the same
 context as the phones
  extension number, in this [local]
 
 
 Additionally, it may not be the first 'exten ='
 line, at least in
 some versions, so best to put them at the end of the
 context.
 
 PLUS: Avoid SIP registrations with a minus '-' in
 them as this breaks
 on several versions.
 
 Hope that helps,
 Steve
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[Asterisk-Users] Hint priority

[richard Coco]

Hi all,

i am desperating, trying to configure an OptiPoint410
with the hint priority.

Here what i have...

OptiPoint410std- exten 2001
X-Lite - exten 2002

But unfortunately no LED ON on my OptiPoint410

sip.conf
[2001]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=default
disallow=all
allow=alaw
allow=ulaw

[2002]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=default
disallow=all
allow=alaw
allow=ulaw

extensions.conf
[default]
exten = 2001,hint,SIP/2001
exten = 2002,hint,SIP/2002

[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2001,2,HangUp

exten = 2002,1,Dial(SIP/2002,10,tr)
exten = 2002,2,HangUp

Has anyone managed get hint working with an
OptiPoint4x0.

thx in advance


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[Asterisk-Users] asterisk with Dialogic BRI /2VFD

[richard Coco]

Hi all,

i have an Asterisk box with an Eicon 4BRI with
chan_capi-cm and every thing works fine. We now plan
to install a new Asterisk using a Dialogic BRI/2VFD.
Is the Dialogic card supported and can i use
chan_capi-cm? Has anyone managed to install this card?
Unfortunately i was unable to find documentation about
Asterisk with Dialogic?

thx in advance for your input!!!

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[Asterisk-Users] AMILogin and case sensitive

[richard Coco]

Hi list,

i am playing around with asterisk manager interface
(and astriskjava) and i notice that the login is not
case sensitive.

so i can use

username: admin
secret: admin
---
# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Asterisk Call Manager/1.0
username: admin
secret: admin
action: login

Response: Success
Message: Authentication accepted

action: logoff

Response: Goodbye
Message: Thanks for all the fish.

Connection closed by foreign host.
---

or
username: Admin
secret: Admin
---
# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Asterisk Call Manager/1.0
username: Admin
secret: Admin
action: login

Response: Success
Message: Authentication accepted

action: logoff

Response: Goodbye
Message: Thanks for all the fish.
---

is that correct?

thx in advance



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Re: [Asterisk-Users] Asterisk and Hipath interconnections

[richard Coco]

Hi,

if yo are looking a way to interconnect Asterisk with
a HiPath 4000 via IP, so you have 2 ways to do it.

- via oh323 (for HiPath 4000 version 1 and 2)
- since HiPath4000 version 3 you are able to
interconnect using sipQ (SIP Trunking)



--- Viktor Tatianin [EMAIL PROTECTED] wrote:

 
 Hello
 
 Can anyone know where may download chan_cornet for
 interconnection Asterisk
 and Hipath via IP
 
 Thanks
 
 Viktor
 
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Re: [Asterisk-Users] Asterisk and Hipath interconnections

[richard Coco]

Hi again,

i don't think that the HiPath2000 is an Asterisk based
system. AFAIK the HiPath2K is only configurable using
a Web-based tool (no console access). For the moment
the HiPath2K will only be release with CornetIP (HFA).
No SIP (panned in a second step) and unfortunazely no
IAX are avalaible.

so if teh HiPath2K is an Asterisk based PBX, it meens
that Siemens has developped a pseudo chan_cornet...
but i don't think so...

--- Tele Cost Price Reducer [EMAIL PROTECTED] wrote:

 hi all,
 maybe i am mistaken but it seems to me that the
 HiPath 2000 series is an
 Asterisk based system.
 why am i saying this? because Siemens announce it is
 a Linux, Open Source
 system.
 so, as i do not know any OTHER PBX Linux- Open
 Source system rather then
 Asterisk, does anybody know something else?
 otherwise, if it is an Asterisk system, so why there
 is a need for Cornet?
 you can interconnect with IAX, isn't it?
 
 Mickey
 
 On 2/27/06, Stephen Arulraj
 [EMAIL PROTECTED] wrote:
 
  Hi Victor
 
  Looking for the same answers here too. We are
 regional distributors for
  Hicom HiPath in this part of the world and until
 now we are still
  waiting for chan_cornet to come around. So far we
 have successfully
  interconnected via BRI (mISDN) and PRI (Zaptel)
 and it works great.
 
  Let's see if it's too good to be true soon.
 
  Best regards,
  Stephen
 
  Viktor Tatianin wrote:
 
  Hello
  
  Can anyone know where may download chan_cornet
 for interconnection
  Asterisk
  and Hipath via IP
  
  Thanks
  
  Viktor
  
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Re: [Asterisk-Users] chan_capi-cm and DID

[richard Coco]

Hi armin,

thx for the answer. I have connected the BRI on a
HiPazt4000 and i still have the same issue. So i think
i have a problem with my ISDN line. I will contact my
provider. May be a reset of the line will solve the
problem.

rich.

--- Armin Schindler [EMAIL PROTECTED] wrote:

 On Tue, 17 Jan 2006, richard Coco wrote:
  Hi Armin,
  
  thx for your feedback, but what do you mean with
 Did
  you load the card with config for DID on that
 port?
  
  I have loaded the modules with:
  modprobe capi 
  modprobe kernelcapi 
  modprobe divacapi 
  modprobe divas
  
  and then loaded divactrl like this:
  divactrl load -f ETSI
  
  I suppose that this is ok (it works without did)?
 Or
  have i forgotten something?
 
 With 
   divactrl load -f ETSI
 you load the card to PtMP (which is the default) on
 all four ports.
 Use
   divactrl load -c 1 -SeparateConfig -u1
 where the '1' of -u1 means second port.
 E.g. -u is first port, -u1 -u2 -u3 is port 2,3,4.
 
 When using -SeparateConfig, the X-extension is
 available
 for many options.
 
 E.g., you can put port 3 and 4 into NT-mode, or even
 run another protocol
 (1TR6, JAPAN, QSIG,...) on other ports.
 
 See
   divactrl load -h
 for all options.
  
 Armin
 
  thx in advance..
  
  --- Armin Schindler [EMAIL PROTECTED] wrote:
  
   On Mon, 16 Jan 2006, richard Coco wrote:
Hi all,

i have asterisk 1.0.9 with an Eicon Diva 4bri
 and
chan_capi-cm-0.6. I have 2 NTBAs (one with did
 and
   one
without).
When using the one without did, i am able to
 place
outgoing and incoming calls. When i use the
 NTBAs
   with
did i have a layer 2 error.

Anyone an idea?
   
   Did you load the card with config for DID on
 that
   port?
   What are your divactrl parameters? (Or do you
 use
   Eicon Package with xml based config?)
   
   Armin 

-- Executing Dial(SIP/2004-9634,
CAPI/g1/43XX) in new stack
data = g1/43XX
parsed dialstring: 'g1' '43XX' ''
capi request group = 2
parsed dialstring: 'g1' '43XX' ''
  == EICON: Call CAPI/EICON/43XX-6  
   (pres=0x00,
ton=0x00)
CONNECT_REQ ID=001 #0x000c LEN=0065
  Controller/PLCI/NCCI= 0x1
  CIPValue= 0x10
  CalledPartyNumber   =
 8043XX
  CallingPartyNumber  = 00 80
22EyeBeam22 3c20043e
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
  BC  = default
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= 00 00
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

-- Called g1/43XX
CONNECT_CONF ID=001 #0x000c LEN=0014
  Controller/PLCI/NCCI= 0x201
  Info= 0x0

-- EICON: received CONNECT_CONF PLCI =
 0x201
DISCONNECT_IND ID=001 #0x0011 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Reason  = 0x3302

DISCONNECT_RESP ID=001 #0x0011 LEN=0012
  Controller/PLCI/NCCI= 0x201

CAPI INFO 0x3302: Protocol error
 layer 2
  == EICON: CAPI Hangingup
  == EICON: Interface cleanup PLCI=0x201
  == No one is available to answer at this
 time

my capi.conf looks like:
[DID]
controller=1,2,3,4
isdnmode=did
incomingmsn=*
softdtmf=on
relaxdtmf=on
accountcode=
context=DID
echocancel=yes
;echocancelold=yes
devices=2
group=1



   
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Re: [Asterisk-Users] chan_capi-cm and DID

[richard Coco]

Hi Armin,

thx for your feedback, but what do you mean with Did
you load the card with config for DID on that port?

I have loaded the modules with:
modprobe capi 
modprobe kernelcapi 
modprobe divacapi 
modprobe divas

and then loaded divactrl like this:
divactrl load -f ETSI

I suppose that this is ok (it works without did)? Or
have i forgotten something?

thx in advance..

--- Armin Schindler [EMAIL PROTECTED] wrote:

 On Mon, 16 Jan 2006, richard Coco wrote:
  Hi all,
  
  i have asterisk 1.0.9 with an Eicon Diva 4bri and
  chan_capi-cm-0.6. I have 2 NTBAs (one with did and
 one
  without).
  When using the one without did, i am able to place
  outgoing and incoming calls. When i use the NTBAs
 with
  did i have a layer 2 error.
  
  Anyone an idea?
 
 Did you load the card with config for DID on that
 port?
 What are your divactrl parameters? (Or do you use
 Eicon Package with xml based config?)
 
 Armin 
  
  -- Executing Dial(SIP/2004-9634,
  CAPI/g1/43XX) in new stack
  data = g1/43XX
  parsed dialstring: 'g1' '43XX' ''
  capi request group = 2
  parsed dialstring: 'g1' '43XX' ''
== EICON: Call CAPI/EICON/43XX-6  
 (pres=0x00,
  ton=0x00)
  CONNECT_REQ ID=001 #0x000c LEN=0065
Controller/PLCI/NCCI= 0x1
CIPValue= 0x10
CalledPartyNumber   = 8043XX
CallingPartyNumber  = 00 80
  22EyeBeam22 3c20043e
CalledPartySubaddress   = default
CallingPartySubaddress  = default
BProtocol
 B1protocol = 0x1
 B2protocol = 0x1
 B3protocol = 0x0
 B1configuration= default
 B2configuration= default
 B3configuration= default
BC  = default
LLC = default
HLC = default
AdditionalInfo
 BChannelinformation= 00 00
 Keypadfacility = default
 Useruserdata   = default
 Facilitydataarray  = default
  
  -- Called g1/43XX
  CONNECT_CONF ID=001 #0x000c LEN=0014
Controller/PLCI/NCCI= 0x201
Info= 0x0
  
  -- EICON: received CONNECT_CONF PLCI = 0x201
  DISCONNECT_IND ID=001 #0x0011 LEN=0014
Controller/PLCI/NCCI= 0x201
Reason  = 0x3302
  
  DISCONNECT_RESP ID=001 #0x0011 LEN=0012
Controller/PLCI/NCCI= 0x201
  
  CAPI INFO 0x3302: Protocol error layer 2
== EICON: CAPI Hangingup
== EICON: Interface cleanup PLCI=0x201
== No one is available to answer at this time
  
  my capi.conf looks like:
  [DID]
  controller=1,2,3,4
  isdnmode=did
  incomingmsn=*
  softdtmf=on
  relaxdtmf=on
  accountcode=
  context=DID
  echocancel=yes
  ;echocancelold=yes
  devices=2
  group=1
  
  
  
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[Asterisk-Users] chan_capi-cm and DID

[richard Coco]

Hi all,

i have asterisk 1.0.9 with an Eicon Diva 4bri and
chan_capi-cm-0.6. I have 2 NTBAs (one with did and one
without).
When using the one without did, i am able to place
outgoing and incoming calls. When i use the NTBAs with
did i have a layer 2 error.

Anyone an idea?


-- Executing Dial(SIP/2004-9634,
CAPI/g1/43XX) in new stack
data = g1/43XX
parsed dialstring: 'g1' '43XX' ''
capi request group = 2
parsed dialstring: 'g1' '43XX' ''
  == EICON: Call CAPI/EICON/43XX-6   (pres=0x00,
ton=0x00)
CONNECT_REQ ID=001 #0x000c LEN=0065
  Controller/PLCI/NCCI= 0x1
  CIPValue= 0x10
  CalledPartyNumber   = 8043XX
  CallingPartyNumber  = 00 80
22EyeBeam22 3c20043e
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
  BC  = default
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= 00 00
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default

-- Called g1/43XX
CONNECT_CONF ID=001 #0x000c LEN=0014
  Controller/PLCI/NCCI= 0x201
  Info= 0x0

-- EICON: received CONNECT_CONF PLCI = 0x201
DISCONNECT_IND ID=001 #0x0011 LEN=0014
  Controller/PLCI/NCCI= 0x201
  Reason  = 0x3302

DISCONNECT_RESP ID=001 #0x0011 LEN=0012
  Controller/PLCI/NCCI= 0x201

CAPI INFO 0x3302: Protocol error layer 2
  == EICON: CAPI Hangingup
  == EICON: Interface cleanup PLCI=0x201
  == No one is available to answer at this time

my capi.conf looks like:
[DID]
controller=1,2,3,4
isdnmode=did
incomingmsn=*
softdtmf=on
relaxdtmf=on
accountcode=
context=DID
echocancel=yes
;echocancelold=yes
devices=2
group=1



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Re: [Asterisk-Users] [offtopic] Asterisk -IP- Siemens HiPath 4000

[richard Coco]

Hi,

we have interconnected Asterisk with a HiPath4000 V1.0
using a H.323 Trunk. You have to install the oh323
channel from [1]. On your HiPath4000 V1.0 or V2.0 you
need a HG3550 board for IP-Trunking.
If you have the version 3.0 then the HiPath supports
SIP-Trunking but i have not tested it yet.

otherwise you can interconnect Asterisk with HiPath
using ISDN. I have in addition to the h323 Trunk a
backup trunk using a Diva Eicon 4 Bri (chan_capi) and
it works fine too. I suppose it will work with Digium
cards too.

[1]http://www.inaccessnetworks.com/projects/asterisk-oh323



--- Dmitry Ivanov [EMAIL PROTECTED] wrote:

 Hello!
 
 Is it possible to connect Siemens HiPath 4000 to
 Asterisk? What 
 equipment required on Siemens side? I mean IP not
 E1.
 
 Sorry for asking here. Siemens-related websites use
 salesperson 
 language. There is no technical information.
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[Asterisk-Users] CallParking and chan_capi-cm-0.6

[richard Coco]

Hi all,

i run into problems using park calling with chan_capi.
My setup looks like this

[200X]--[Asterisk]--[PSTN]

For internal calls [1] and for incoming call from
PSTN[2] every thing works fine. Unfortunately when a
sip extension (say 2007) makes an outgoing call to
PSTN and 2007 tranfers to extension 700, asterisk
doesn't tell him the park position. In the CLI you can
only see that asterisk plays digits/7 [3]. However
if e.g extension 200x calls into extension 701 he is
able to take the parked call.

any suggestions? thx in advance

[3]
  -- Started music on hold, class 'default', on
CAPI/EICON/4349768000-3d
  == Parked CAPI/EICON/4349768000-3d on 701. Will
timeout back to incoming,4349768000,1 in 60 seconds
-- Playing 'digits/7' (language 'en')
-- Added extension '701' priority 1 to parkedcalls
-- Executing ParkedCall(SIP/2008-bd95, 701) in
new stack
-- Stopped music on hold on
CAPI/EICON/4349768000-3d
-- Channel SIP/2008-bd95 connected to parked call
701
  == EICON: CAPI Hangingup
  == Spawn extension (international, 701, 1) exited
non-zero on 'SIP/2008-bd95'
CAPI INFO 0x3490: Normal call clearing


[1]
 -- Executing Dial(SIP/2007-e62a,
SIP/2006|10|TtHhr) in new stack
-- Called 2006
-- SIP/2006-84ac is ringing
-- SIP/2006-84ac answered SIP/2007-e62a
-- Attempting native bridge of SIP/2007-e62a and
SIP/2006-84ac
-- Started music on hold, class 'default', on
SIP/2007-e62a
-- Playing 'pbx-transfer' (language 'en')
-- Stopped music on hold on SIP/2007-e62a
-- Started music on hold, class 'default', on
SIP/2007-e62a
  == Parked SIP/2007-e62a on 701. Will timeout back to
international,2006,1 in 6
0 seconds
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Added extension '701' priority 1 to parkedcalls
  == Spawn extension (international, 2006, 1) exited
KEEPALIVE on 'SIP/2007-e62a
'
-- Executing ParkedCall(SIP/2008-4e32, 701) in
new stack
-- Stopped music on hold on SIP/2007-e62a
-- Channel SIP/2008-4e32 connected to parked call
701
-- Attempting native bridge of SIP/2008-4e32 and
SIP/2007-e62a
  == Spawn extension (international, 701, 1) exited
non-zero on 'SIP/2008-4e32'


[2]
-- Executing Dial(CAPI/EICON/434xxx-3c,
SIP/2007|10|TtHhr) in new stack
-- Called 2007
-- SIP/2007-8d67 is ringing
-- SIP/2007-8d67 answered CAPI/EICON/434xxx-3c
-- Started music on hold, class 'default', on
CAPI/EICON/434xxx-3c
-- Playing 'pbx-transfer' (language 'en')
-- Stopped music on hold on CAPI/EICON/434xxx-3c
-- Started music on hold, class 'default', on
CAPI/EICON/434xxx-3c
  == Parked CAPI/EICON/434xxx-3c on 701. Will timeout
back to local,2007,1 in 60 seconds
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Added extension '701' priority 1 to parkedcalls
  == Spawn extension (local, 2007, 1) exited KEEPALIVE
on 'CAPI/EICON/434xxx-3c'
-- Executing ParkedCall(SIP/2008-58dd, 701) in
new stack
-- Stopped music on hold on CAPI/EICON/434xxx-3c
-- Channel SIP/2008-58dd connected to parked call
701
  == EICON: CAPI Hangingup
  == Spawn extension (international, 701, 1) exited
non-zero on 'SIP/2008-58dd'
CAPI INFO 0x3490: Normal call clearing



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[Asterisk-Users] moh on optipoint400

[richard Coco]

Hi all,

i'm wondering if anyone has ever managed to get moh
working on Siemens OptiPoint400?

if yes, can you please explain how you did it...

thx.




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[Asterisk-Users] Asterisk and DrayTek Vigor2600VGi

[richard Coco]

Hi all,


I'm trying to configure a remote user with a DrayTek
2600Vgi. The setup looks like this.

[SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone]

I can place calls to the DrayTek and recieve calls
from the analog phone. However, the calling party does
not hear the 
called party (one way audio). The audio for the remote
user works fine. VPN works fine too and i have no
drops in my fw-logs.

my sip.conf looks like this. (ext 2005 is my DrayTek
and ext 2006 is the local sip user)

[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes

[2006]
type=friend
callerid=OptiPoint600 2006
context=international
host=dynamic
user=2006
secret=x
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw

[2005]
type=friend
canreinvite=no
host=dynamic
user=2005
secret=x
dtmfmode=rfc2833
callerid=Draytek 2005
context=international
disallow=all
allow=alaw
allow=ulaw

i try to play around with externip, localnet, nat and
canreinvite but still have the same issue.
Sporadically i can see a Maximum retries exceeded on
call ... for seqno 102 (Non-critical Request on the
CLI.
I have also replaced the analog phone with a
softclient on my laptop connected behind the vigor,
but have the same problem.

Has anybody managed to get a similar setup running?

any ideas, suggestions are wellcome... thx in
advance...

richard.



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Re: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi

[richard Coco]

Hi Alessio



[SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone]
 
 I tried a similar setup some times ago and it was
 working, have you
 put the private ip address of the asterisk box in
 the vigor setup ?
 
 Can you ping the private address of the vigor from
 the asterisk box
 and viceversa ?

I am able to ping the private addr of the vigor from *
and of couse viceversa. The vigor setup seems to be ok
(vpn is up and *sip show peers* shows that the vigor
is registred.). I can also call from and to Asterisk,
so the signalisation is ok. I have only problem with
RTP packets (one way audio)

anyhow thx for the feedback...




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Re: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi

[richard Coco]

Alessio, Sergio

 So an upgrade is of course necessary.

i have upgraded the vigor. Bad news... i am not able
to register the draytek anymore. But using a XLite on
my pc behind the Vigor works now fine (no one way
audio).

however i have an other question. I saw you put for
the bindaddr same thing like 192.168.0.3. Is that the
ip addr from your Asterisk?

i will sniff and look wat happens...

ps: Sergio, sorry for the mail... bad reply..



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[Asterisk-Users] Asterisk connected with CAPI

[richard Coco]

Hi all,

i'm trying to install a EICON DIVA 4BRI (on CentOS 4.1
 2.6.9-22.0.1.EL)  using latest package from
sourceforge (chan_capi-cm-0.6.tar.gz).
I have installed divactrl_2.1.tar.gz and untared
protocols_all.tar.bz2 in /usr/share/eicon.
---
lsmod gives me the following...
Module  Size  Used by
divacapi  157937  0
capi   18177  0
capifs  5961  2 capi
kernelcapi 44641  2 divacapi,capi
md5 4033  1
ipv6  234881  12
lp 12077  0
autofs423237  0
i2c_dev11329  0
i2c_core   22081  1 i2c_dev
sunrpc159269  1
microcode   6881  0
button  6481  0
battery 8901  0
ac  4805  0
uhci_hcd   31065  0
parport_pc 24577  0
parport37129  2 lp,parport_pc
divas  76345  0
divadidd   13081  2 divacapi,divas
e100   41793  0
mii 4673  1 e100
floppy 58481  0
dm_snapshot16901  0
dm_zero 2369  0
dm_mirror  27825  0
ext3  116809  2
jbd71385  1 ext3
dm_mod 56661  6
dm_snapshot,dm_zero,dm_mirror
---
   


Starting divactrl
---
[EMAIL PROTECTED] asterisk]# divactrl load -c 1 -f ETSI
Start adapter Nr:1 - 'Diva Server 4BRI-8M 2.0 PCI',
SN: 7113 ... OK
[EMAIL PROTECTED] asterisk]#


but the /var/log/asterisk/messages gives me following
errors when i try to start asterisk:
Nov  4 12:25:45 WARNING[2658]: CAPI not installed,
CAPI disabled!
Nov  4 12:25:45 WARNING[2658]: chan_capi.so:
load_module failed, returning -1
Nov  4 12:25:45 WARNING[2658]: Loading module
chan_capi.so failed!

Is CAPI really not installed or have i forgotten
something? Here my capi.conf and modules.conf
;
; CAPI config
;
;

; general section

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8


[EICON]
controller=1,2,3,4
isdnmode=msn
incomingmsn=*
softdtmf=on
relaxdtmf=on
accountcode=
context=incoming
echocancel=yes
devices=2
group=1

;
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_musiconhold.so
load = chan_capi.so
noload = chan_alsa.so
[global]
chan_modem.so=yes
chan_capi.so=yes


thx in advance



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Re: [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server

[richard Coco]

Hi Jacky,

thx for the feedback

rich.

--- Jacky [EMAIL PROTECTED] wrote:

 Hi, Richard,
 
 I still try, but fail with rtp transfer.
 
 
 2005/9/27, richard Coco [EMAIL PROTECTED]:
 
   I still find out how to let LCS 2005 accept SIP
   invite from Asterisk,
   Need more help.
 
  Hi jacky,
 
  can you please share your experience and explain
 how
  to let LCS accept SIP invite from Asterisk.
 
  I deseperate trying to place a call from asterisk
 to
  LCS. (calling from Asterisk to LCS using
 TCP_SUPPORT
  seems to work fine)
 
  thx in advance
 
 
   2005/8/13, bubuk [EMAIL PROTECTED]:
Hi,
   
I already posted this in the user list, but
 this
   list is probably the
better one.
   
My question was: Does anyone played around
 with
   the LCS and Asterisk?
Because the LCS is doing no RFC compliant SIP,
 i
   wonder if it can work.
Google couldn't tell me. If someon heard about
   that, please let me know.
   
Thank you
Volker
   
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Re: [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server

[richard Coco]

 I still find out how to let LCS 2005 accept SIP
 invite from Asterisk,
 Need more help.

Hi jacky,

can you please share your experience and explain how
to let LCS accept SIP invite from Asterisk.

I deseperate trying to place a call from asterisk to
LCS. (calling from Asterisk to LCS using TCP_SUPPORT
seems to work fine)

thx in advance

 
 2005/8/13, bubuk [EMAIL PROTECTED]:
  Hi,
  
  I already posted this in the user list, but this
 list is probably the
  better one.
  
  My question was: Does anyone played around with
 the LCS and Asterisk?
  Because the LCS is doing no RFC compliant SIP, i
 wonder if it can work.
  Google couldn't tell me. If someon heard about
 that, please let me know.
  
  Thank you
  Volker
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Re: [Asterisk-Users] MS Live Communication Server

[richard Coco]

Hi all,

i have now managed to place a call from LCS to
asterisk/pstn and it seems to work fine. Unfortunately
i have still problems for incomming calls from
asterisk/pstn to LCS.

i have seen in the mailinglist that there seems to be
problem calling from lcs to asterisk. Have anyone
maneged to place a call from lcs to *.

thx in advance... 

--- richard Coco [EMAIL PROTECTED] wrote:

 
 Hi,
 
 i have the same setup too.
 

[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
 
 Unfortunately i don't know how to configure the
 dialplan in my LCS. Can you please give me a hint
 where to configure this.
 
 thx.
 
 
 --- Jacky [EMAIL PROTECTED] wrote:
 
  LCS 2005 just support SIP TCP or TLS right now.
  so you must patch asterisk chan_sip.c support TCP,
  look http://bugs.digium.com/view.php?id=4903
  
  I have successful call to asterisk's SIP peer or
  PSTN use Office
  Communicator 2005(sign-in my LCS 2005)
  but I can't use Dial(SIP/[EMAIL PROTECTED]) ,
 let
  asterisk's SIP user invite
  LCS's user.
  
  Need any input.
  
  
  2005/8/11, bubuk [EMAIL PROTECTED]:
   Hi List!
   
   does anyone played around with the LCS and
  Asterisk? Because the LCS is
   doing no RFC compliant SIP, i wonder if it can
  work. Google couldn't
   tell me. If someon heared about that, please let
  me know.
   
   The fact i figured out is that the Border
  Controler from Jasomi can be
   used as a gateway from MS-LCS-SIP to regular
 SIP.
  But that is not really
   handy and expensive too.
   
   Thank you
   Volker
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Re: [Asterisk-Users] MS Live Communication Server

[richard Coco]

Hi,

i have the same setup too.

[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]

Unfortunately i don't know how to configure the
dialplan in my LCS. Can you please give me a hint
where to configure this.

thx.


--- Jacky [EMAIL PROTECTED] wrote:

 LCS 2005 just support SIP TCP or TLS right now.
 so you must patch asterisk chan_sip.c support TCP,
 look http://bugs.digium.com/view.php?id=4903
 
 I have successful call to asterisk's SIP peer or
 PSTN use Office
 Communicator 2005(sign-in my LCS 2005)
 but I can't use Dial(SIP/[EMAIL PROTECTED]) , let
 asterisk's SIP user invite
 LCS's user.
 
 Need any input.
 
 
 2005/8/11, bubuk [EMAIL PROTECTED]:
  Hi List!
  
  does anyone played around with the LCS and
 Asterisk? Because the LCS is
  doing no RFC compliant SIP, i wonder if it can
 work. Google couldn't
  tell me. If someon heared about that, please let
 me know.
  
  The fact i figured out is that the Border
 Controler from Jasomi can be
  used as a gateway from MS-LCS-SIP to regular SIP.
 But that is not really
  handy and expensive too.
  
  Thank you
  Volker
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Re: [Asterisk-Users] Optipoint 600 Cant boot - development shell active

[richard Coco]

Hi,

The only thing i know is that you need a netbootserver
using five special files. So, if possible, ask Siemens
for the optipoint 600 netboot upgrade procedure. AFAIK
it is a known problem...

hope it helps...

--- Anthony Cox [EMAIL PROTECTED] wrote:

 Not strictly a problem with Asterisk but one of my
 phones.  A couple of days 
 ago I decided to update the firmware in my Optipoint
 600 Office which looked 
 as though it went swimmingly until that is, it
 rebooted.
 
 Since then the phone just boots up and displays the
 following:
 Can't Boot!!
 Development shell active.
 
 It doesn't try to request a DHCP address, in fact it
 does seem to do anything 
 on the network and the key pad does nothing.
 
 Can anyone suggest a remedy?  Anyone know how to get
 the development shell to 
 do something?
 
 Thanks in advance.
 Anthony.
 
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Re: [Asterisk-Users] connecting Asterisk with Siemens HiPath4000

[richard Coco]

 I already have OH323 support in Asterisk, but have
 no clue how to
 configure the HiPath.

hi...

oh323 is the only thing you need for Astersik. For the
HiPath it depends on which version you have.

FOR HiPath4000 V1.0
---
for version 1.0 you need a HG3550 V1.1 Board.
-Configure the Board using AMO BCSU
Then load the FW into the HG3550. The following steps
are necessary (all via Boot CLI/Console connected to
HG3550 via V.24 interface with a normal Terminal).
-Format Flash
-Load a boot control file
-Load SW Image into flash
-Use the Wizard for the initial setup.
After that the HG3550 is READY and you have to
configure the initial setup using the Routing-Wizard


FOR HiPath4000 V2.0
---
For the v2.0 you need a HG3550 V2.0.Unfortunately
there is no Wizard for the initial setup anymore, so
you have to configure every thing with AMO.

The configuration of the HG3550 is IMHO difficult and
without support from Siemens nearly impossible.

hope it helps...



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[Asterisk-Users] Large installation with Asterisk

[richard Coco]

Hi all,

i am looking for informations about large installation
with Asterisk (~3000 users). Has anybody experience
with such a setup. Any comments, suggestions or
problems would be appreciated.

thx in advance...



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Re: [Asterisk-Users] HiPath 4000 and Asterisk

[richard Coco]

--- [EMAIL PROTECTED] wrote:
 I'm trying to setup Asterisk trunk to Siemens HiPath
 4000 V2.01

i suppose you mean version 2.0 ;-)

 What would be the best way to do so? I am a bit
 confused because as far
 as I've understand this PBX doesn't support H323,
 but I saw somewhere
 someone who created a cornet trunk and it worked
 using H323.

I have a HiPath4000 V1.0 interconnected to Asterisk
using a STMI board (HG3550) and oh323. The
interoperability works well. The chan_cornet AFAIK is
not released by Steffen. The interconnection between
H4kV2.0 and * is identical, use a HG3550 V2.0 for the
H4k and oh323 for *. 



 I've read some information about the cornet
 connectivity which is in
 development - does anyone knows the status of that?

AFAIK not released :-(


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Re: [Asterisk-Users] Re: Siemens optiPoint 420 phone and Asterisk

[richard Coco]

Hi Franz,

ok, can you please inform me (the list) if the Optipoint 420 with the firmware 4.0.22A work with Asterisk. If so i will try to contact our contact at Siemens and organize some Optipoint 420.

chan_cornet, which would support the proprietary Siemens protocol, is notpart of the CVS tree yet (thanks to Steffen Koepf for writing this).BTW Have you additional information about Steffen's chan_cornet. Is there a beta version of the chan_cornet available for testing. As mentioned in my first post we use for the moment oh.323 and i'm very intersted to testit if possible.

thx in advance...Franz Knipp [EMAIL PROTECTED] wrote:
Dear Richard,On Fri, 15 Apr 2005, [EMAIL PROTECTED] wrote: The latest firmware for optipoint420 advance SIP seems to be version 4.0.22A, released for HiPath8000.thanks for this information. I've contacted my customer adviser atSiemens, he'll try to organize me this version. What siemens PBX do you use?It's a HiPath 3300 (Rack version) with the extension containing 4 ISDNports to connect to *. I don't know... maybe it will work... We only have several OptiPoint400 and they work fine.The risk of making the phone unuseable by installing a wrong firmwareseems too high for me, so I won't try that.Thanks for the help!Bye,Franz___Asterisk-Users mailing
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Re: [Asterisk-Users] Siemens optiPoint 420 phone and Asterisk

[richard Coco]

Franz Knipp [EMAIL PROTECTED] wrote:

Hi,today I've got two Siemens optiPoint 420 phones and I want to connectthem to an existing Asterisk server.I didn't find any SIP firmware for that phone, according to announcementsit will be released later this year (hopefully soon).
The latest firmware for optipoint420 advance SIP seems to be version 4.0.22A, released for HiPath8000. Unfortunately on the Siemens page theonlySIP image that can be downloadedis for OptiPoint400 (www.hipath.de then -download - software/version 2.3.14). For Optipoint410/420 only the HFA version is available.
chan_cornet, which would support the proprietary Siemens protocol, is notpart of the CVS tree yet (thanks to Steffen Koepf for writing this).
Is Steffen's chan_cornet available for testing? We have * connected to a HiPath4x00 using oh323. 
Maybe, someone of you can help me getting this phones working withAsterisk by pointing out a good starting point for my investigation andown development (if necessary) ;-)Last but not least, some kind of network diagram to clarify the situation:ISDN NT ---[Siemens PBX]--(S0)--[Asterisk]--(IP)--[optiPoint]
What siemens PBX do you use?
Does anybody know, if it is worth trying out the optiPoint 400 SIPfirmware on the 410/420 phones?
I don't know... maybe it will work... We only have several OptiPoint400 and they work fine.

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[Asterisk-Users] voicemail access

[richard Coco]

Hi,

my setup
[pbx]---[oh323]--[asterisk]

calling from the pbx into the voicemail gives following outputin the console

-- Executing VoiceMailMain("OH323/R1909", "") in new stackApr 5 19:05:46 DEBUG[11862037]: res_adsi.c:212 __adsi_transmit_messages: No ADSI CPE detected (0) -- Playing 'vm-login' (language 'en')Apr 5 19:05:51 WARNING[11862037]: app_voicemail.c:3238 vm_execmain: Couldn't read username
you can see that Asterisk plays the vm-login but the calling party (from pbx) doesn't hear anything. What is the message *NO ADSI CPE detected*?
thx in advance...




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[Asterisk-Users] SoftClient for Pocket PC

[richard Coco]

Hi List,

Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens)an register itwith asterisk?

any suggestions?

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Re: [Asterisk-Users] chan_cornet

[richard Coco]

Hi,

we use the oh323 driver (see the post from Joao for installationhttp://www.archivum.info/asterisk-users@lists.digium.com/2005-01/msg00931.html). 

in the oh323.conf
[general]
listenPort= 192.x.x.x /the ip @ of the HG3550
fastStart=yes /*enable fast start
context=voip-h323
codec=G711A

in the extensions.conf

exten = _0.,2,Dial,OH323/h323:[EMAIL PROTECTED],tr /* for outgoing to HiPath
[voip-h323]
include = default /* for incoming from HiPath.
the rest are default settings.
The H4K installation is more difficult. You have to use a HG3550 board.
1. install HG3550 board
2. initial installation of the loadware (use the latest version!)
3. use WebManagment to configure the routing to Asterisk.

marek cervenka [EMAIL PROTECTED] wrote:
 i agree... no H.323 support for endpoint (HG3530) but you still have h.323 (for the moment only version 2.0) support for ip-trunking (HG3550). So what if you have the following setup. [OPTIPOINT400_HFA]--[HIPAT4K][oh.323][ASTERISK]--[OPTIPOINT400_SIP].how can i configure ip-trunking from HI4K to asterisk?any example h323 conf for asterisk?---Marek CervenkaCentrum Vypocetni TechnikyCVT - http://cvt.fpf.slu.czFPF SLU OPAVA - http://www.fpf.slu.czLCNA - http://lcna.slu.cz===___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options
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Re: [Asterisk-Users] chan_cornet

[richard Coco]

Hi,

i agree... no H.323 support for endpoint (HG3530) but you still have h.323 (for the moment only version 2.0)support for ip-trunking (HG3550).
So what if you have the following setup. 

[OPTIPOINT400_HFA]--[HIPAT4K][oh.323][ASTERISK]--[OPTIPOINT400_SIP].

Am i right when i suppose that the chan_cornet will replacethe oh.323.

[OPTIPOINT400_HFA]--[HIPAT4K][chan_cornet][ASTERISK]--[OPTIPOINT400_SIP].
Steffen Koepf [EMAIL PROTECTED] wrote:
Hello, but what do you mean with *new hipath version doesn't  support H.323 anymore*? What version are you talking about?  As far as i know the new version of HiPath4000 V2.0 still  supports H.323 (STMI2).HiPath 3000 - H.323 SupportHiPath 4000 - NO H.323 Support and nothing else but cornet (voip).cornet-ip is basic h.323 + addons, but it does notwork with standard h.323 devices.cu,Steffen___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] chan_cornet

[richard Coco]

GIBERT Frédéric [EMAIL PROTECTED] wrote:

STLS4 is a 4 BRI ports card to connect to carrier.
STMD8 is a card to connect 8 ISDN Siemens phones (optiset)

STMD8 is not a board for Optisets. You have to use a SLMO/SLU board to register an Optiset/Optipoint500.
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Re: [Asterisk-Users] chan_cornet

[richard Coco]

Hi,

The HG1500 is a HiPath3000 board and i don't have experience with Asterisk and HiPath3K.
What we have is an Asterisk connected to a Siemens HiPath4000 over a H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens HG3550 only supports H.323 V2.0 (so not a lot of features are available). May be Steffen's chan_cornet will change this.
Are there any news about this project?Joao Pereira [EMAIL PROTECTED] wrote:
Hibut did anyone have ever used a Siemens HiPath PBX with Asterisk?If you made it, please tell me how...I read that chan_cornet does exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs there any Digium Hardware solution for the Asterisk HiPath connection?ThanksJoao Pereira- Original Message -From: "Luís Palma" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"Sent: Tuesday, January 04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500 Hi, It doesn't tell you much but it looks like that you are not alone when trying to integrate with Siemens Hicom. It seems someone has decided to make it by himself.
 http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 -, Joao Pereira <[EMAIL PROTECTED]>wrote:  Hi  I want to know the best way to connect Asterisk to a Siemens HiPathHG1500  PBX. Until now I came out with 3 solutions:   1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs  Siemens licences and Digium hardware)  2-Asterisk connecting to the PSTN phones with Voice Modems (goodideia!!!  but its analog... doesnt have caller information...)  3-Using RDIS interfaces to connect the Siemens PBX   does someone have other ideias?   Thanks  Joao Pereira   ___  Asterisk-Users mailing list 
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Re: [Asterisk-Users] chan_cornet

[richard Coco]

What do you mean with *But doesnt Digium have Asterisk-HiPath solutions?*. If you are meaning a connection with Digium cards so...sorry i've never usedDigium cards. Buti thought (see your first thread *connect Asterisk with Siemens HiPath HG1500*), you are looking for a way to connectAsterisk to the HG.If you use HG1500 you have to configure a h323 channel (h.323 or oh.323). If not, you can try toconfigurechan_capi and try to connect Asterisk (e.g with an EICONDiva card)to a STLS4 (for HiPath3500) or a STMD8(forHiPath3700).
hope it will help...
if in a few days you have additional informations about chan_cornet, please let me (the list) know.

thx
Joao Pereira [EMAIL PROTECTED] wrote:




Hi
I dont knowif Steffen's chan_cornet is working. I emailed him, but with no result.
Yesterday I read this article
http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom

It has some solutions... but not yet a direct Asterisk-HiPath connection.

But doesnt Digium have Asterisk-HiPath solutions?

Joao

- Original Message - 
From: richard Coco 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Sent: Wednesday, January 05, 2005 12:13 PM
Subject: Re: [Asterisk-Users] chan_cornet 

Hi,

The HG1500 is a HiPath3000 board and i don't have experience with Asterisk and HiPath3K.
What we have is an Asterisk connected to a Siemens HiPath4000 over a H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens HG3550 only supports H.323 V2.0 (so not a lot of features are available). May be Steffen's chan_cornet will change this.
Are there any news about this project?Joao Pereira [EMAIL PROTECTED] wrote:
Hibut did anyone have ever used a Siemens HiPath PBX with Asterisk?If you made it, please tell me how...I read that chan_cornet does exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs there any Digium Hardware solution for the Asterisk HiPath connection?ThanksJoao Pereira- Original Message -From: "Luís Palma" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<ASTERISK-USERS@LISTS.DIGIUM.COM>Sent: Tuesday, January 04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500 Hi, It doesn't tell you much but it looks like that you are not alone when trying to integrate with Siemens Hicom. It seems someone has decided to make it by himself.
 http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 -, Joao Pereira <[EMAIL PROTECTED]>wrote:  Hi  I want to know the best way to connect Asterisk to a Siemens HiPathHG1500  PBX. Until now I came out with 3 solutions:   1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs  Siemens licences and Digium hardware)  2-Asterisk connecting to the PSTN phones with Voice Modems (goodideia!!!  but its analog... doesnt have caller information...)  3-Using RDIS interfaces to connect the Siemens PBX   does someone have other ideias?   Thanks  Joao Pereira   ___  Asterisk-Users mailing list 
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Re: [Asterisk-Users] chan_cornet

[richard Coco]

Hello Steffen,

hey it sounds very good...!!! but what do you mean with *new hipath version doesn't support H.323 anymore*? What version are you talking about? As far as i know the new version of HiPath4000 V2.0 still supports H.323 (STMI2).Steffen Koepf [EMAIL PROTECTED] wrote:
Hello, I dont know if Steffen's chan_cornet is working. I emailed him, but with no result.You are not patient enough ;)You got an answer one minute ago ;)No it is not ready, it is work in progress.At the moment i'm forced to get a Asterisk-SMS Gateway working here,with our old PBX, but i hope it works soon so that i can proceed withchan_cornet. At the moment, Optipoint 400 and 600s can register tothe chan_cornet, and one can call them so that they ring. There is somelittle work to be done, to get the voice working (a bit H.323 stuff),and with a little editor (for entering the numbers, the phone can't handle this), the Phone2PBX part should work. And then the next goalis the PBX2PBX stuff.That newer HiPath PBXs are worse, coz Siemens dropped the H.323-Support,that means they do not support one stand
 ard voip
 protocol. They saidthat they will support SIP in the future, but they say this for morethan a year now. That means, one can connect now that PBXs with TDM-Lines(S2M, BRI) or to another cornet-ip supporting device like IPDAs (smallersiemens pbxs that connect to a main PBX) or PBXs and hopefully, chan_cornetsometime ;). cu,Steffen___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Music on Hold

[richard Coco]

Hi all,

i'm trying to configure MoH for OptiPoint400std SIP, but it doesn't work. If i try with a softclient moh works fine. Somebody experience with OptiPoint400? Is there additional settings on the Optipoint?

Any help would be much appreciated!!
thx.
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RE: [Asterisk-Users] Music on Hold

[richard Coco]

Hi Erik,

thx for the feedback. I turned off the Silence Suppression but unfortunately it doesn't work.
(i don't think that throwing it out of the window will resolve my problem... but it will think about it... to be continued;-))"E. Versaevel" [EMAIL PROTECTED] wrote:







If you’re using G.711 make sure you’ve got silence suppression turned OF, seems that the phone only receives rtp while sending it.
(or, do as we did, throw the OptiPoint out of the window J)







Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens richard CocoVerzonden: maandag 27 december 2004 15:31Aan: asterisk-users@lists.digium.comOnderwerp: [Asterisk-Users] Music on Hold


Hi all,



i'm trying to configure MoH for OptiPoint400std SIP, but it doesn't work. If i try with a softclient moh works fine. Somebody experience with OptiPoint400? Is there additional settings on the Optipoint?



Any help would be much appreciated!!

thx.



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Re: [Asterisk-Users] mail function

[richard Coco]

hi 

check the voicemail.conf

Attach=yes
Attach causes Asterisk to copy a voicemail message to an audio file and send it to the user as an attachment in an e-mail voicemail notification message. The default is not to do this. Attach takes two values yes or no. 

extension_number = voicemail_password,user_name,user_email_address,user_pager_email_address,user_option(s) 


Tasos Daskalopoulos [EMAIL PROTECTED] wrote:




Hello there

Can Asterisk send me a Mail wenn a voice mail arrived for me ?
if yes how can i do this ?

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Re: [Asterisk-Users] Connecting Siemens HiCom PBX with Asterisk through E1

[richard Coco]

richard Coco [EMAIL PROTECTED] wrote:

Peter Svensson [EMAIL PROTECTED] wrote:


On Sun, 19 Dec 2004, Jens Kübler wrote: I've bought the Wildcard TE110 some days ago but I'm unable to get it to work  with Siemens HiCom 300.  I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4  and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard  takes a few seconds and sets the link to green (OK).It should be 1 to 4 and 2 to 5, not 3. The pairs are 1-2 and 4-5.
Hi,
Using a DIUS2 on the Hicom, you have to loop 3-10 and 7-14..
Using a DIUN2 on the Hicom, the loop is not necessary.

BTW 1-9 is Tx and 7-14 is RX.

sorry... 1-9 Tx and 8-15 Rx (not 7-14)

hope it will help.
 2. I've tried to connect our running E1 line from the telco with wildcard. The  modem (modulates HDSL to G703 120 Ohms). I've used a 1:1 cable that did not  work. I even tried to connect the copper wires by hand which resulted that  the modem gave me a green power light but Wildcard stayed on a waving red  light.That should have worked, as long as 1-1, 2-2, 4-4 and 5-5. Are you sure the signalling is correct? What did zttool say? 3. I have plugged out our running PBX and connected it to Wildcard which  resulted in a green light for one second and then the state from zttool  switched to yellow (and Wildcard to constant red light).Yellow alert is remote alert, right? That would indicate that the path from the pbx to asterisk is ok, but not the path from asterisk to the pbx. The protocol used by the modem is hdb3 and /etc/zaptel.conf is adjusted  according to
 this.Do you have crc4 enabled? Care to post your zaptel.conf.I think you have to start asterisk on the span to get the upper layers enabled.Peter___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [Asterisk-Users] Connecting Siemens HiCom PBX with Asterisk through E1

[richard Coco]

Peter Svensson [EMAIL PROTECTED] wrote:


On Sun, 19 Dec 2004, Jens Kübler wrote: I've bought the Wildcard TE110 some days ago but I'm unable to get it to work  with Siemens HiCom 300.  I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4  and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard  takes a few seconds and sets the link to green (OK).It should be 1 to 4 and 2 to 5, not 3. The pairs are 1-2 and 4-5.
Hi,
Using a DIUS2 on the Hicom, you have to loop 3-10 and 7-14..
Using a DIUN2 on the Hicom, the loop is not necessary.

BTW 1-9 is Tx and 7-14 is RX.

hope it will help.
 2. I've tried to connect our running E1 line from the telco with wildcard. The  modem (modulates HDSL to G703 120 Ohms). I've used a 1:1 cable that did not  work. I even tried to connect the copper wires by hand which resulted that  the modem gave me a green power light but Wildcard stayed on a waving red  light.That should have worked, as long as 1-1, 2-2, 4-4 and 5-5. Are you sure the signalling is correct? What did zttool say? 3. I have plugged out our running PBX and connected it to Wildcard which  resulted in a green light for one second and then the state from zttool  switched to yellow (and Wildcard to constant red light).Yellow alert is remote alert, right? That would indicate that the path from the pbx to asterisk is ok, but not the path from asterisk to the pbx. The protocol used by the modem is hdb3 and /etc/zaptel.conf is adjusted  according to
 this.Do you have crc4 enabled? Care to post your zaptel.conf.I think you have to start asterisk on the span to get the upper layers enabled.Peter___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Display on OptiPoint400std SIP

[richard Coco]

Hi all,

I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient.

Any ideas? Thx!

sip.conf
[2005]
type=friend
callerid="OptiPoint" 2005
context=default
host=dynamic
disallow=all
allow=ulaw
allow=alaw

extensions.conf
exten = 2005,1,Dial(SIP/${EXTEN},10,tr)
exten = 2005,2,Congestion


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[Asterisk-Users] [oh323] sporadic call setup

[richard Coco]

Hi all,

this is my actuel setup

[SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900]

Linux CentOS 3.3 (2.4.21-20.EL.c0)
asterisk-1.0.1
asterisk-oh323-0.6.3b
openh323_1.12.2
pwlib_1.5.2

Calling from SIPphone to the extension 8900 works always. 
Calling from 8900 to SIPphone works only sporadicly without any recognizeable pattern. 
Find below the output of the debug command:  asterisk -vvvcdg
*CLI Dec 13 12:36:19 DEBUG[-1220658256]: chan_oh323.c:3218 cleanup_h323_connection: ENTER cleanup_h323_connection.Dec 13 12:36:19 WARNING[-1220658256]: chan_oh323.c:3232 cleanup_h323_connection: Call ip$192.168.204.130:2024/26923 not found.Dec 13 12:36:19 WARNING[-1220658256]: chan_oh323.c:3232 cleanup_h323_connection: Call ip$192.168.204.130:2024/26923 not found.Dec 13 12:36:19 DEBUG[-1220658256]: chan_oh323.c:3234 cleanup_h323_connection: LEAVE cleanup_h323_connection.Segmentation fault (core dumped)

*CLI 0:59.372 H225 Answer:997cc90 PWLib Assertion fail: Null pointer reference, file /usr/src/openh323/src/h225_1.cxx, line 360, Error=22
Abort, Core dump, Ignore?

and here a debug output from a successfull call setup

*CLI Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:3381 init_h323_connection: ENTER init_h323_connection.Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2201 new_oh323: Player fds 32,33 - Recorder fds 34,35 - Event pipe 36,37.Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:3445 init_h323_connection: Created new call in 0.Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2678 copy_call_details: --- CALL DETAILS ---Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2679 copy_call_details: call_token = 'ip$192.168.204.130:2206/27015'Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2680 copy_call_details: call_source_alias =
 'Ü¨Ü , 9205,0721*8 []'Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2681 copy_call_details: call_dest_alias = '2005 'Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2682 copy_call_details: call_source_e164 = '8900'Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2683 copy_call_details: call_dest_e164 = '2005'Dec 13 13:13:05 DEBUG[-1275274320]:
 chan_oh323.c:2684 copy_call_details: remote_app = ' 118.67/30593'Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2685 copy_call_details: remote_addr = '192.168.204.130'Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2686 copy_call_details: local_addr = '192.168.204.223'Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2334 ast_oh323_new: OH323/R27015: Raw format set to ALAW.Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2423 ast_oh323_new: Context is 'voip-h323', extension is '2005'.Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:2426 ast_oh323_new: CallerID/ANI is
 '"Ü¨Ü , 9205,0721*8 " 8900'.Dec 13 13:13:05 DEBUG[-1296254032]: pbx.c:1260 pbx_extension_helper: Launching 'Dial'Dec 13 13:13:05 DEBUG[-1296254032]: app_dial.c:490 dial_exec: SIMPLE DIAL (NO URL)Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:2395 sip_alloc: Allocating new SIP call for (null)Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1295 create_addr: Setting NAT on RTP to 0Urgent handlerDec 13 13:13:05
 DEBUG[-1296254032]: chan_sip.c:1536 sip_call: Outgoing Call for 2005Dec 13 13:13:05 DEBUG[-1296254032]: chan_sip.c:1669 update_user_counter: Call from user '2005' is 1 out of 0 -- Called 2005Urgent handlerDec 13 13:13:05 DEBUG[-1296254032]: chan_oh323.c:1136 oh323_indicate: OH323/R27015: Indicating condition 3.Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:3460 init_h323_connection: OH323/R27015: Channel created and attached.Dec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:3462 init_h323_connection: NEW STATE: NULL -- RINGDec 13 13:13:05 DEBUG[-1275274320]: chan_oh323.c:3492 init_h323_connection: LEAVE init_h323_connection.Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:3027 setup_h323_connection: ENTER setup_h323_connection.Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:3034 setup_h323_connection: PLAYER.Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:3078 setup_h323_connection: Request to open an existing 
 channel
 0 with no established direction.Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2678 copy_call_details: --- CALL DETAILS ---Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2679 copy_call_details: call_token = 'ip$192.168.204.130:2206/27015'Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2680 copy_call_details: call_source_alias = 'Ü¨Ü , 9205,0721*8 []'Dec 13 13:13:05 DEBUG[-1285764176]:
 chan_oh323.c:2681 copy_call_details: call_dest_alias = '2005 'Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2682 copy_call_details: call_source_e164 = '8900'Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2683 copy_call_details: call_dest_e164 = '2005'Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2684 copy_call_details: remote_app = ' 118.67/30593'Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2685 copy_call_details: remote_addr = '192.168.204.130'Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:2686 copy_call_details: local_addr = 

[Asterisk-Users] outgoing calls

[richard Coco]

Hi,
here what i have:
[2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004]
Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 onRedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL.

Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following error,
-- Executing Dial("SIP/2001-8a8e", "Modem/ttyI0:998004|20|r") in new stackOct 11 13:49:12 WARNING[262159]: channel.c:1901 ast_request: No channel type registered for 'Modem'Oct 11 13:49:12 NOTICE[262159]: app_dial.c:742 dial_exec: Unable to create channel of type 'Modem' == Everyone is busy/congested at this timeOct 11 13:49:22 WARNING[262159]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'default'

Extension 2001 gives "unreachable"99 is thecode using for outgoing calls. 
;sip.conf[2001]type=friendsecret=2001auth=2001callerid="user 2001" 2001host=dynamicdisallow=allcontext=defaultallow=ulawallow=alaw
;extensions.conf[default]exten = 2001,1,NoOp( call for ${EXTEN})exten = 2001,2,Dial(SIP/${EXTEN},60,tr)exten = 2001,3,Congestionexten = _99.,1,Dial(Modem/ttyI0:${EXTEN:0},20,r)
;modem.conf[interfaces]context=remotedriver=i4llanguage=entype=autodetectdialtype=tonemode=ringdevice = /dev/ttyI0

Have I missed something in my extensions.conf? or in modem.conf?
thanks for your support...
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