Re: [asterisk-users] Server response time

[Gordon Henderson]

On Mon, 1 Mar 2010, Juan C. Villa wrote:

> On 2/28/2010 10:21 AM, Gordon Henderson wrote:
>> On Sun, 28 Feb 2010, Juan C. Villa wrote:
>>
>>> Hey Guys,
>>>
>>> I am considering leasing a new server in Germany to run my Asterisk
>>> infrastructure and I was wondering how response time would affect the
>>> performance of the system. Right now I have a response time of around
>>> 60-70ms with my server in California. The server in Germany would have a
>>> response time of around 140ms (both ways). My DID/Termination providers
>>> are in Canada and the USA, and all my voip boxes are also in the USA.
>>> Any suggestions or recommendations?
>>>
>> Being based in the UK, I'd say why not the UK rather then Germany - we're
>> closer to the US after-all :)
>>
>> However, one thing we don't know: Where are you and your customers based?
>>
>> I also find it odd that a lot of people UK based still think they can get
>> better deals (cheaper&  more b/w) by hosting in the US rather than in the
>> UK - so I'm curious as to why you'd want to host outside the US...
>>
>> But as long as you're not passing media then anywhere you have good
>> connectivity ought to work - however if you are passing media, then I'd be
>> concerned that someone in California is calling their neighbour and the
>> data is going all the way to Germany and back again... That really will be
>> noticeable...
>>
>
> In response to Gordon: Hetzner offers the best dedicated server deal I
> have every seen. I have been a Cari.net client for  over a year now, but
> I am needing a more powerful server and I don't want to pay $200+ a
> month for it. Hetzner has a connection to the Level 3 network that
> recently installed a transoceanic fiber optic link with a lag of less
> than 40 ms.

You're not going to get much better than 40ms each way from NY to Europe 
because as Scotty would say: Ye canny break the laws o' physics! (Actually 
light in fibre takes 26.1ms according to Wolfram alpha but London to NY 
has been ~40ms each way since as long as I've been involved with that 
stuff (mid 90's)

And most big ISPs in europe now connect to Level3 - e.g. the co-lo I use 
in deepest darkest england (nowhere near London, although we do have Gb to 
London) has a ping time like:

   gordon @ unicorn: ping -q -c10 www.nyiix.net
   PING ns3.nyiix.net (209.137.140.21) 56(84) bytes of data.

   --- ns3.nyiix.net ping statistics ---
   10 packets transmitted, 10 received, 0% packet loss, time 9009ms
   rtt min/avg/max/mdev = 78.865/79.947/86.082/2.079 ms

It goes via L3 and that's probably not the best end-point, but it's close 
enough, and ~40ms each way.

> The total lag from Germany to USA (2 way) is around ~110ms (Just tested
> it today). Who this cause any issues with my VoIP applications? Right
> now I have two VoIP boxes installed in Switzerland which are connected
> to my server in California (avg response time = 190ms) and I have no
> problems at all. What would you guys advice?

So are you passing data, or just signalling? If data, then why? (Although 
I guess you're actually terminating to the PSTN in those countries?) But 
as you already have servers in Switzerland, why can't you use those to run 
some extended tests, and work it out for yourself?

Personally, I'd not even think about servers in another country unless I 
had good reason to - and good "remote hands"/support, etc. and a 
requirement to plumb in to the local PSTN - either directly or via a local 
VoIP carrier - and even then, if it's via a local VoIP carrier - why not 
just connect directly to them from 'home' rather than put a box over 
there.

But I if you already have servers in .ch which you indicate you're happy 
with, then I guess you do have good reason to have them there, so since 
.de is just up the road from .ch, then if you're happy with the ISP/co-lo 
then go for it...

Do make sure the facility has multiple carrier ISPs though - if L3 does go 
down (and no-ones perfect), you still need a way to get to it - L3 isn't 
the only backhaul ISP with trans-atlantic links - get the co-lo's AS 
number and see who they're peering with using the various 'whois' tools, 
etc.

Good luck!

Gordon

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Re: [asterisk-users] Does Asterisk 1.6.2.1 Support SIP TLS encryption

[Klaus Darilion]

Am 02.03.2010 07:26, schrieb Zhang Shukun:
> hi, all
>
> i want to realize more secure communication between asterisk sip end users.
>
> so i want to know Does Asterisk 1.6.2.1 Support SIP TLS encryption?

yes. But Asterisk does not support SRTP. Thus, only the SIP signaling is 
encrypted, not the audio/video streams.


> if you can tell me same specific example to do encrypt, it's very appreciated.

see configs/sip.conf.sample

>
> Thanks!
>

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Re: [asterisk-users] SIP Trunk with "multiple" remote ip-addresses

[Klaus Darilion]

Am 02.03.2010 08:50, schrieb Magnus Benngård:
> Hi,
>
> Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No
> problem to get outgoing calls to work but i have some problems with
> incoming.
>
> Did set "srvlookup=yes" in sip.conf. "Sending" all outgoing calls to
> "sip-corporate.tele2.se" which is either sip-corporate1.tele2.se
> (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46).
>
> If i do a "sip show peer Tele2", I see that Asterisk has chosen one of
> them: ToHost : sip-corporate.tele2.se
> Addr->IP : 130.244.190.46 Port 5060
>
> Now my problems starts, when Tele2 sends a call to my Asterisk, the call
> can come frome any of those two ip-adresses. If it comes from
> 130.244.190.46 everything if fine, but if it comes from 130.244.190.42:
> "[Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite:
> Failed to authenticate!"
>
> I thought "srvlookup=yes" should take care about that, but then i read a
> little bit more and found: "Note: Asterisk only uses the first host in
> SRV records". :(

Hi Magnus!

Asterisk does not support multiple SRV records (expcet there were some 
recent changes which I missed) - it takes one of the most priors and use 
it all the time.

Thus, in your scenario you have to specify the possible inbound sources 
manually as peers:

[tele2-1]
type=peer
host=130.244.190.42
context=fromTele2
...
[tele2-2]
type=peer
host=130.244.190.46
context=fromTele2
...


regards
klaus


>
> Can anyone plz give me some hint howto solve my problem?
>
> Regards,
>
> Magnus
>

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Re: [asterisk-users] rtcachefriends & qualify

[jonas kellens]

Thank you for your answer, Nic.

It seems that by putting rtcachefriend=yes, the qualify works as
expected and even changes made to my realtime MySQL-DB take affect
immediately without the need of a reload (I changed the username and
name).

However the old username and name are still valuable and using this old
SIP-user, one can still make outgoing calls. Receiving calls is no
longer possible :

WARNING[32439]: app_dial.c:1272 dial_exec_full: Unable to create channel
of type 'SIP' (cause 20 - Unknown)

Adding 'rtautoclear=yes' to sip.conf makes no difference. Changes to
SIP-account are taken immediately, but the old SIP-credentials are still
valid. (even after an unregister and re-register)

Only after a "sip reload" I get the notice :

[Mar  2 10:41:03] NOTICE[32498]: chan_sip.c:15889
handle_request_register: Registration from
'"Gerrie"' failed for
'192.168.1.105' - No matching peer found

So a "sip reload" is always necessary to clear the cache ??


Jonas.

On Mon, 2010-03-01 at 14:31 +, Nic Colledge wrote:
> Hi,
> 
>  
> 
> I think so, maybe someone can help clarify this for me also. I have:
> 
> rtcachefriends=yes
> 
> rtautoclear=yes
> 
> in sip.conf and was under the impression that this caches the settings
> from the database until a user unregisters. When they unregister the
> data is removed from the cache (rtautoclear). For me this was a nice
> compromise.
> 
>  
> 
> This is from memory but I’m pretty sure I got this from the
> documentation online, if someone can confirm what I’m saying that
> would be sweet.
> 
>  
> 
> Thanks.
> 
> Nic.

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Re: [asterisk-users] MeetMe and usernum

[Tony Mountifield]

In article <3de056a31003010645x2c4481fbr5b05923d88614...@mail.gmail.com>,
David Backeberg  wrote:
> On Mon, Mar 1, 2010 at 6:42 AM, Emrah  wrote:
> > I am trying to get the usernum of a user when dialing in to a MeetMe
> > conference. Is there somehow a possibility to save the usernum of a
> > MeetMe participant into a variable? Everything should be done through
> > the DialPlan, no manager and no *cli.
> 
> You don't say what version you're running.
> 
> I second Steve's claim. Even with 1.6, I can't think of how to do what
> you want without resorting to AGI. Which is technically in the
> dialplan, but you're going to have to do extra work elsewhere.
> 
> If you're using 1.6, you will enjoy knowing about 'meetme list 
> concise', which you can then process with awk.
> 
> If you absolutely don't want to do AGI, you could always modify
> meetme.c, recompile, and share your work with others. I think you'll
> find that harder than writing an AGI.

Actually it would be fairly simple to add a variable set in the
appropriate part of app_meetme.c:

{
  char temp[12];
  snprintf(tmp, sizeof(tmp), "%d", user->user_no);
  pbx_builtin_setvar_helper(chan, "MEETME_USERNUM", tmp);
}

The problem becomes: how would you use this variable? You can't
execute dialplan statements based on it until the user returns from
the call to MeetMe. That would be fine for logging, but not for
much else.

I think you need to describe your original problem, i.e. why you think
you need the user number. There may be other ways to achieve what you
want.

Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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[asterisk-users] Echo cancellation on DAHDI

[DHAVAL INDRODIYA]

Dear All,

How can we know the On board supports echo cancellation

I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
02)*board

all working fine but sometimes i got echo when user are calling a PRI.

is there any way to know on board echo cancellation .


regards

Dhaval
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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[jonas kellens]

I'd like to add to my thread that realtime SIP peers do not seem to be
surviving a "sip reload".

step 1 : 2 realtime SIP peers are registered to Asterisk, they can make
a phone call to each other.
step 2 : I do a 'sip reload'
step 3 : the 2 realtime SIP peers are no longer able to phone to each
other 

[Mar  2 11:32:41] WARNING[32668]: app_dial.c:1272 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
[Mar  2 11:32:41]   == Everyone is busy/congested at this time (1:0/0/1)
[Mar  2 11:32:41]   == Auto fallthrough, channel
'SIP/gerrie001-09ed70d0' status is 'CHANUNAVAIL'

I look at the mysql-table 'sip_buddies' and the values for 'ipaddr' and
'port' are still filled in and correct.

When executing 'sip show peers', the realtime peers also have
disappeared.
At first there was :
Name/username  HostDyn Nat ACL Port Status
Realtime  
gerrie002/gerrie002192.168.1.104D   N  5060 OK
(10 ms) Cached RT 
gerrie001/gerrie001192.168.1.105D   N  5060 OK
(30 ms) Cached RT

Now there is :
Name/username  HostDyn Nat ACL Port Status
Realtime  
gerrie002/gerrie002192.168.1.104 D   N  5060
UNREACHABLE Cached RT 

Using Zoiper softphone, the SIP-accounts still show status 'registered'.

Re-registering is the only thing that helps :
Name/username  HostDyn Nat ACL Port Status
Realtime  
gerrie001/gerrie001192.168.1.105D   N  5060 OK
(9 ms)  Cached RT 
gerrie002(Unspecified)D   N  0
UNREACHABLE Cached RT 

And for account 2 :
Name/username  HostDyn Nat ACL Port Status
Realtime  
gerrie002/gerrie002192.168.1.104D   N  5060 OK
(6 ms)  Cached RT 
gerrie001/gerrie001192.168.1.105D   N  5060 OK
(9 ms)  Cached RT 

In the mysql-DB, the field 'regseconds' turns from zero to some large
integer...

I can reproduce the above very easy by just initiating 'sip reload'...

Is this behaviour normal ??

Jonas.
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Re: [asterisk-users] rtcachefriends & qualify

[Mindaugas Kezys]

The problems we have with Asterisk Realtime:

 

   1. After reload all registrations are void.

   2. Without reload prune does not take effect.

 

Test it in your scenario also.

 

Regards,

Mindaugas Kezys

 

Kolmisoft UAB 

VoIP Billing Solutions

e-mail: i...@kolmisoft.com

URL: http://www.kolmisoft.com

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Tuesday, March 02, 2010 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] rtcachefriends & qualify

 

Thank you for your answer, Nic.

It seems that by putting rtcachefriend=yes, the qualify works as expected and 
even changes made to my realtime MySQL-DB take affect immediately without the 
need of a reload (I changed the username and name).

However the old username and name are still valuable and using this old 
SIP-user, one can still make outgoing calls. Receiving calls is no longer 
possible :

WARNING[32439]: app_dial.c:1272 dial_exec_full: Unable to create channel of 
type 'SIP' (cause 20 - Unknown)

Adding 'rtautoclear=yes' to sip.conf makes no difference. Changes to 
SIP-account are taken immediately, but the old SIP-credentials are still valid. 
(even after an unregister and re-register)

Only after a "sip reload" I get the notice :

[Mar  2 10:41:03] NOTICE[32498]: chan_sip.c:15889 handle_request_register: 
Registration from '"Gerrie"' failed 
for '192.168.1.105' - No matching peer found

So a "sip reload" is always necessary to clear the cache ??


Jonas.

On Mon, 2010-03-01 at 14:31 +, Nic Colledge wrote: 

Hi,

 

I think so, maybe someone can help clarify this for me also. I have:

rtcachefriends=yes

rtautoclear=yes

in sip.conf and was under the impression that this caches the settings from the 
database until a user unregisters. When they unregister the data is removed 
from the cache (rtautoclear). For me this was a nice compromise.

 

This is from memory but I’m pretty sure I got this from the documentation 
online, if someone can confirm what I’m saying that would be sweet.

 

Thanks.

Nic. 

 
 
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[asterisk-users] 1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why?

[Kristijan Vrban]

Asterisk 1.4.29

BLF-SUBSCRIBE go to internal IP (ngrep output):

U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060
  SUBSCRIBE sip:1...@62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From:
;tag=vyx8c0trgx..To:
  ;tag=as13e7cb7c..Call-ID:
3c2768d8487f-rbzdwjzdbgcs..CSeq: 1163 SUBSCRIBE..Contact:
;reg-id=1..max-forwards: 70.
  .event: dialog..user-agent: snom320/8.2.25..expires: 60..Accept:
application/dialog-info+xml..Content-Length: 0

U 2010/03/02 11:34:06.053870 192.168.4.109:5060 -> 192.168.55.31:2048
  SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport;received=212.78.xxx.xxx..From:
;tag=vyx8c0
  trgx..To: ;tag=as13e7cb7c..Call-ID:
3c2768d8487f-rbzdwjzdbgcs..CSeq: 1163 SUBSCRIBE..User-Agent: asterisk
1.4.29..Allow: INVITE, ACK, CANCEL, OPTIONS,
  BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported:
replaces..WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="5a8a1268"..Content-Length: 0



But VM-SUBSCRIBE go to external IP (ngrep output):

U 2010/03/02 11:33:46.362857 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060
  SUBSCRIBE sip:aster...@62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-lubj3r12xmcy;rport..From:
;tag=bov99lxeez
  ..To: ;tag=as586c72f7..Call-ID:
3c2670215123-ymlw0ru3an2r..CSeq: 851 SUBSCRIBE..Contact:
;reg-id=1..max-f
  orwards: 70..event: message-summary..user-agent:
snom320/8.2.25..expires: 60..Accept:
application/simple-message-summary..Content-Length: 0

U 2010/03/02 11:33:46.363003 62.134.xxx.xxx:5060 -> 212.78.xxx.xxx:2048
  SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-lubj3r12xmcy;rport;received=212.78.xxx.xxx..From:
;tag=bov99l
  xeez..To: ;tag=as586c72f7..Call-ID:
3c2670215123-ymlw0ru3an2r..CSeq: 851 SUBSCRIBE..User-Agent: asterisk
1.4.29..Allow: INVITE, ACK, CANCEL, OPT
  IONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported:
replaces..WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="66b3b8ff"..Content-Length: 0


Why? Look's like a bug for me?


sip show peer K922002626:

  * Name   : K922002626
  Realtime peer: Yes, cached
  Secret   : 
  MD5Secret: 
  Context  : K9220
  Subscr.Cont. : 
  Language : de
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : 3...@k9220
  VM Extension : voicemail
  LastMsgsSent : 0/0
  Call limit   : 5
  Dynamic  : Yes
  Callerid : "" <>
  MaxCallBR: 384 kbps
  Expire   : 232
  Insecure : no
  Nat  : No
  ACL  : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 212.78.xxx.xxx Port 2048
  Defaddr->IP  : 0.0.0.0 Port 2048
  Def. Username: K922002626
  SIP Options  : (none)
  Codecs   : 0x80e (gsm|ulaw|alaw|g726)
  Codec Order  : (alaw:20,ulaw:20,g726:20,gsm:20)
  Auto-Framing:  No
  Status   : OK (47 ms)
  Useragent: snom320/8.2.25
  Reg. Contact : sip:k922002...@212.78.xxx.xxx:2048

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[asterisk-users] Sip module problem

[Luis Silva]

Hi,

I need some help debugging a sip situation.

I started to have problems with sip trunks, using more than one trunk (and
sometimes using only one) the sip module seems to freeze.

My local extensions lost registration and also the trunks.  The only way
that I can restart the sip is removing the trunks. If I make sip reload or
restart asterisk the sip module takes many many time before starting.

I use the logger in full in order to debug the problem but don't see
anything strange. 

 

During the freeze If I make sip show peers , all the extensions and trunks
are unreachable, (where etx 10 and 11 and trunk telepac5)

 

12/12  (Unspecified)D   N  0UNKNOWN

11/11  172.16.1.100 D   N  5063 UNREACHABLE

100(Unspecified)D   N  0UNKNOWN

10/10  172.16.1.101 D   N  25124UNREACHABLE

telepac5/+351302028197 213.13.89.675060 UNREACHABLE

 

But I know that are request's coming to my box if I check with netstat I'm
receiving packages  

 

[r...@localhost ~]# netstat -an|grep 5060

udp0  0 0.0.0.0:50600.0.0.0:*

[r...@localhost ~]# netstat -an|grep 5060

udp 1840  0 0.0.0.0:50600.0.0.0:*

[r...@localhost ~]# netstat -an|grep 5060

udp 1840  0 0.0.0.0:50600.0.0.0:*

[r...@localhost ~]# netstat -an|grep 5060

udp 1840  0 0.0.0.0:50600.0.0.0:*

[r...@localhost ~]# netstat -an|grep 5060

udp27600  0 0.0.0.0:50600.0.0.0:*

[r...@localhost ~]# netstat -an|grep 5060

udp33120  0 0.0.0.0:50600.0.0.0:*

 

The second field  Recv-Q is according to man "The count of bytes not copied
by the user program connected to this socket.", so this means that asterisk
is not getting this packets right?...

 

 I started with asterisk version 1.4.26.1, upgraded to 1.4.29 but got the
same result

 

How can I debug this situation? Is there a way to debug the "insides" of the
sip module?

 

Regards

Luis

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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[Ishfaq Malik]

jonas kellens wrote:
> I'd like to add to my thread that realtime SIP peers do not seem to be 
> surviving a "sip reload".
>
> step 1 : 2 realtime SIP peers are registered to Asterisk, they can 
> make a phone call to each other.
> step 2 : I do a 'sip reload'
> step 3 : the 2 realtime SIP peers are no longer able to phone to each 
> other
>
> [Mar  2 11:32:41] WARNING[32668]: app_dial.c:1272 dial_exec_full: 
> Unable to create channel of type 'SIP' (cause 20 - Unknown)
> [Mar  2 11:32:41]   == Everyone is busy/congested at this time (1:0/0/1)
> [Mar  2 11:32:41]   == Auto fallthrough, channel 
> 'SIP/gerrie001-09ed70d0' status is 'CHANUNAVAIL'
>
> I look at the mysql-table 'sip_buddies' and the values for 'ipaddr' 
> and 'port' are still filled in and correct.
>
> When executing 'sip show peers', the realtime peers also have disappeared.
> At first there was :
> Name/username  HostDyn Nat ACL Port 
> Status Realtime 
> gerrie002/gerrie002192.168.1.104D   N  5060 OK 
> (10 ms) Cached RT
> gerrie001/gerrie001192.168.1.105D   N  5060 OK 
> (30 ms) Cached RT
>
> Now there is :
> Name/username  HostDyn Nat ACL Port 
> Status Realtime 
> gerrie002/gerrie002192.168.1.104 D   N  5060 
> UNREACHABLE Cached RT
>
> Using Zoiper softphone, the SIP-accounts still show status 'registered'.
>
> Re-registering is the only thing that helps :
> Name/username  HostDyn Nat ACL Port 
> Status Realtime 
> gerrie001/gerrie001192.168.1.105D   N  5060 OK 
> (9 ms)  Cached RT
> gerrie002(Unspecified)D   N  0
> UNREACHABLE Cached RT
>
> And for account 2 :
> Name/username  HostDyn Nat ACL Port 
> Status Realtime 
> gerrie002/gerrie002192.168.1.104D   N  5060 OK 
> (6 ms)  Cached RT
> gerrie001/gerrie001192.168.1.105D   N  5060 OK 
> (9 ms)  Cached RT
>
> In the mysql-DB, the field 'regseconds' turns from zero to some large 
> integer...
>
> I can reproduce the above very easy by just initiating 'sip reload'...
>
> Is this behaviour normal ??
>
> Jonas. 
Hi

In my experience, yes, that is normal behaviour. Generally any SIP phone 
will try to reconnect with the server within 2 mins anyway.

If you are changing RealTime config in your DB you need to do a sip 
prune realtime either directly from asterisk cli or using AMI. You 
really do not need to do a SIP reload when changing the config of one 
sip extension.

Ish
-- 
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Software Developer
PackNet Ltd

Office:   0161 660 3062

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[asterisk-users] cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5

[Andreas Brodmann]

Hi all,

We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to
any newer releases:

We use the following cli command to feed a wave/mp3 file into an existing
conference on an other serve:
/opt/asterisk/sbin/asterisk -r -x "channel originate
Local/confgongad...@xy_features extension confgongp...@xy_features"

The corresponding extensions.conf part looks like that:
--
[XY_Features]
exten => ConfGongAdmin,1,NoCDR()
exten => ConfGongAdmin,n,Set(TIMEOUT(absolute)=10)
exten => ConfGongAdmin,n,Dial(SIP/12...@server)

exten => ConfGongPlay,1,Answer()
exten => ConfGongPlay,n,Set(TIMEOUT(absolute)=10)
exten => ConfGongPlay,n,Wait(2)
exten => ConfGongPlay,n,Playback(/etc/asterisk/sounds/gong)
---

Until asterisk-1.6.2.0 this worked fine.

With later releases including 1.6.2.5 asterisk does a call to
confgongad...@xy_features but once that stands does not
continue with a call to ConfGongPlay.

Our asterisk system is a pure asterisk installation, no dahdi drivers for
timing, as we don't have zaptel/dahi hardware.

What we basically do is we try to play a sound file into an existing
conference on another server.

We have also tried to do the same thing with the ConfBridge application but
have found so far that ConfBridge only works with
phones, e.g. stations that provide RTP which asterisk can use for timing.
When we try to play a sound file into such a conference
from the same server asterisk won't play anything.

Maybe I am just doing it wrong. Any suggestions or help would be
appreciated.

Andreas
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Re: [asterisk-users] Sip module problem

[Philipp von Klitzing]

Hi!

> I started to have problems with sip trunks, using more than one trunk
> (and sometimes using only one) the sip module seems to freeze... My local
> extensions lost registration and also the trunks. The only way that I
> can restart the sip is removing the trunks...

Have seen this also on different 1.4 versions (1.4.18), created a bug
report and was told this to be a DNS issue. I do not think it is, but
never got this one fully solved.

https://issues.asterisk.org/view.php?id=15139
https://issues.asterisk.org/view.php?id=15052

'This bug is surely a DNS related problem, if you look over /main/dns.c
you'll find:
"Asterisk DNS is synchronus at this time. This means that if your DNS
does not work properly, Asterisk might not start properly or a channel
may lock."
Try to change nameservers in /etc/resolv.conf.'

What distribution & kernel are you using?

> If I make sip reload or restart asterisk the sip module takes many many time
> before starting.

That I didn't notice, had to do a reboot (!).

Philipp


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Re: [asterisk-users] Echo cancellation on DAHDI

[Vinícius Fontes]

- "DHAVAL INDRODIYA"  escreveu:

> Dear All,
> 
> How can we know the On board supports echo cancellation
> 
> I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
> 02) board
> 
> all working fine but sometimes i got echo when user are calling a PRI.
> 
> is there any way to know on board echo cancellation .
> 
> 
> regards
> 
> Dhaval

Do you have an echo cancelling module attached to that board? If so, all you 
need is to set echocancel=yes and echocancelwhenbridged=no on your 
chan_dahdi.conf. If you don't... well you should!

Anyway, you can turn on the echocancelling via software with echocancel=256. I 
strongly recommend using OSLEC in that case. You'll need to patch your DAHDI in 
order to use it, but it's totally worth it.

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Re: [asterisk-users] SIP Trunk with "multiple" remote ip-addresses

[Magnus Benngård]

Hi! 

Did a setup of 2 peers as Klaus suggested, it worked thx! 

Has anyone thought about the possibility to add multiple ip/hosts to
"host="? 

I my case: "host=130.244.190.42,130.244.190.46" or
"host=sip-corporate1.tele2.se,sip-corporate2.tele2.se" 

Step 1 could be to send to the first ip/host and accept from both. 

Step 2 could be "round-robin" send if both are up and alive... 

Btw, did try trunk version, no support for multiple SRV records there.  

Am 02.03.2010 08:50, schrieb Magnus Benngård:
> Hi,
>
> Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No
> problem to get outgoing calls to work but i have some problems with
> incoming.
>
> Did set "srvlookup=yes" in sip.conf. "Sending" all outgoing calls to
> "sip-corporate.tele2.se" which is either sip-corporate1.tele2.se
> (130.244.190.42) or sip-corporate1.tele2.se (130.244.190.46).
>
> If i do a "sip show peer Tele2", I see that Asterisk has chosen one of
> them: ToHost : sip-corporate.tele2.se
> Addr->IP
: 130.244.190.46 Port 5060
>
> Now my problems starts, when Tele2 sends a call to my Asterisk, the call
> can come frome any of those two ip-adresses. If it comes from
> 130.244.190.46 everything if fine, but if it comes from 130.244.190.42:
> "[Mar 2 08:46:03] NOTICE[1372]: chan_sip.c:19167 handle_request_invite:
> Failed to authenticate!"
>
> I thought "srvlookup=yes" should take care about that, but then i read a
> little bit more and found: "Note: Asterisk only uses the first host in
> SRV records". :(

Hi Magnus!

Asterisk does not support multiple SRV records (expcet there were some 
recent changes which I missed) - it takes one of the most priors and use 
it all the time.

Thus, in your scenario you have to specify the possible inbound sources 
manually as peers:

[tele2-1]
type=peer
host=130.244.190.42
context=fromTele2
...
[tele2-2]
type=peer
host=130.244.190.46
context=fromTele2
...

regards
klaus

>
> Can anyone plz give me some hint howto solve my problem?
>
>
Regards,
>
> Magnus
> 

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[asterisk-users] dialplan reload: not working with large dialplans

[Andreas Brodmann]

There is a problem that bothered me for a long time:

Since one of the 1.6.0.x patch releases up until 1.6.2.5 a "dialplan reload"
works only once with a bigger dialplan.
If I issue "dialplan reload" again, it won't do anything. After doing so the
cli won't show responses
to any commands anymore.

So if I have to do another change to the dialplan, I have to stop/start
asterisk.

Did anyone encounter a similar issue?

-Andreas
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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[jonas kellens]

On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:

> In my experience, yes, that is normal behaviour. Generally any SIP phone 
> will try to reconnect with the server within 2 mins anyway.

In the Zoiper softphone, it is set to 3600 seconds... I don't want my
customers have to do a lot of configuration on their softphone.
Can I force the SIP clients to re-register every 5 minutes (a setting in
sip.conf ?) ?? Will this cause a lot of overhead ?

> If you are changing RealTime config in your DB you need to do a sip 
> prune realtime either directly from asterisk cli or using AMI. You 
> really do not need to do a SIP reload when changing the config of one 
> sip extension.

I'm using a php-webGUI to change the sip_buddies table. Is their an easy
php class that facilitates working with AMI (as I have no experience
with AMI) ?


Greetingz,
Jonas.
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Re: [asterisk-users] help!!! Internal extensions not connect

[carem gyssell nieto]

Hi Erik, thanks for your help, I found a solution, but this problem only
happens when my server reboot.

I put permissions in:

chmod 777 /var/lib/asterisk/agi-bin/recirdingcheck ...and
chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi

I think the problem is  FreePBX not Asterisk.

But when I reboot lost permissions againI don't understand.

thanks for your help.

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Re: [asterisk-users] help!!! Internal extensions not connect

[Tzafrir Cohen]

On Tue, Mar 02, 2010 at 08:28:52AM -0500, carem gyssell nieto wrote:
> Hi Erik, thanks for your help, I found a solution, but this problem only
> happens when my server reboot.
> 
> I put permissions in:
> 
> chmod 777 /var/lib/asterisk/agi-bin/recirdingcheck ...and
> chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi

hmod 777 normally means you did something wrong.

I suspect the file had to be executable. In which case 755 (or: a+x)
would have done the job just as well.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] dialplan reload: not working with large dialplans

[Tzafrir Cohen]

On Tue, Mar 02, 2010 at 01:56:37PM +0100, Andreas Brodmann wrote:
> There is a problem that bothered me for a long time:
> 
> Since one of the 1.6.0.x patch releases up until 1.6.2.5 a "dialplan reload"
> works only once with a bigger dialplan.
> If I issue "dialplan reload" again, it won't do anything. After doing so the
> cli won't show responses
> to any commands anymore.
> 
> So if I have to do another change to the dialplan, I have to stop/start
> asterisk.
> 
> Did anyone encounter a similar issue?

Can you post somewhere a dialplan that reproduces this?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] dialplan reload: not working with large dialplans

[Andreas Brodmann]

Hi Tzafrir,

yes, I will have to 'anonymize' the dialplan, is this list the right place
though?

-Andreas

2010/3/2 Tzafrir Cohen 

>  On Tue, Mar 02, 2010 at 01:56:37PM +0100, Andreas Brodmann wrote:
> > There is a problem that bothered me for a long time:
> >
> > Since one of the 1.6.0.x patch releases up until 1.6.2.5 a "dialplan
> reload"
> > works only once with a bigger dialplan.
> > If I issue "dialplan reload" again, it won't do anything. After doing so
> the
> > cli won't show responses
> > to any commands anymore.
> >
> > So if I have to do another change to the dialplan, I have to stop/start
> > asterisk.
> >
> > Did anyone encounter a similar issue?
>
> Can you post somewhere a dialplan that reproduces this?
>
> --
>   Tzafrir Cohen
> icq#16849755  
> jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
> --
> _
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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[Ishfaq Malik]

jonas kellens wrote:
> On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
>> In my experience, yes, that is normal behaviour. Generally any SIP phone 
>> will try to reconnect with the server within 2 mins anyway.
>> 
> In the Zoiper softphone, it is set to 3600 seconds... I don't want my 
> customers have to do a lot of configuration on their softphone.
> Can I force the SIP clients to re-register every 5 minutes (a setting 
> in sip.conf ?) ?? Will this cause a lot of overhead ? 
If you get the AMI part working you will no longer need to do SIP 
reloads and this becomes academic.
>> If you are changing RealTime config in your DB you need to do a sip 
>> prune realtime either directly from asterisk cli or using AMI. You 
>> really do not need to do a SIP reload when changing the config of one 
>> sip extension.
>> 
> I'm using a php-webGUI to change the sip_buddies table. Is their an 
> easy php class that facilitates working with AMI (as I have no 
> experience with AMI) ?
Hi, Have a look at this page

http://www.voip-info.org/wiki/view/Asterisk+manager+API

Further down the page it has links to using the AMI with different 
programming languages including example classes. In the end I ended up 
writing my own class as I was only doing 3 or 4 things with the AMI and 
didn't need the rest.  All you are doing is opening a socket on the 
asterisk machine and writing to and reading from it so it's not exactly 
rocket science.

I went through exactly the problems you are having myself a fair few 
months ago. It took me a day or 2 to get it all sorted out in my test 
environment so it's not too difficult to implement.
>
>
> Greetingz,
> Jonas. 

-- 
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Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] X-Lite won't register

[carem gyssell nieto]

Hi,
It's your S.O firewall disable? It's SeLinux disable? you can see that with
the 'setup' command if you are using some red hat distribution.

If you use 'sip show peers' command in your CLI you can see the sip
peers?if notthe problem is your manager connection between Asterisk
and FreePBX.


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Re: [asterisk-users] Echo cancellation on DAHDI

[Brian]

On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
> - "DHAVAL INDRODIYA"  escreveu:
> 
> > Dear All,
> > 
> > How can we know the On board supports echo cancellation
> > 
> > I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
> > 02) board
> > 
> > all working fine but sometimes i got echo when user are calling a PRI.
> > 
> > is there any way to know on board echo cancellation .
> > 
> > 
> > regards
> > 
> > Dhaval
> 
> Do you have an echo cancelling module attached to that board? If so, all you 
> need is to set echocancel=yes and echocancelwhenbridged=no on your 
> chan_dahdi.conf. If you don't... well you should!
> 
> Anyway, you can turn on the echocancelling via software with echocancel=256. 
> I strongly recommend using OSLEC in that case. You'll need to patch your 
> DAHDI in order to use it, but it's totally worth it.
> 
On the subject of DAHDI -v- OSLEC.

I never had any luck getting it to work with DAHDI 2.2.1 despite
following:

http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi

All I ever go was a bad case of the blues :-(

make[3]: *** No rule to make target
`/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by
`/usr/src/dahdi/linux/drivers/dahdi/echo.o'.

I guess I missed something somewhere???


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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[jonas kellens]

On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: 

> If you are changing RealTime config in your DB you need to do a sip 
> prune realtime either directly from asterisk cli or using AMI. You 
> really do not need to do a SIP reload when changing the config of one 
> sip extension.

I notice that after a "sip prune realtime all" I also loose all of my
realtime sip peers. Same result actually as with "sip reload".

I close the softphone of gerrie2 (becomes unspecified)

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
Realtime  
gerrie005/gerrie005192.168.1.106D   N  5060 OK
(4 ms)  Cached RT 
gerrie002/gerrie002(Unspecified)D   N  0
UNKNOWNCached RT 
gerrie001/gerrie001192.168.1.105D   N  5060 OK
(11 ms) Cached RT

I prune the realtime peers to no longer have gerrie002 in cache :

asterisk*CLI> sip prune realtime all
3 peers pruned.
2 users pruned.
[Mar  2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 91

The realtime peers are all gone :

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
Realtime 

Internal call fails :

[Mar  2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable
to create channel of type 'SIP' (cause 20 - Unknown)
[Mar  2 15:46:38]   == Everyone is busy/congested at this time (1:0/0/1)
[Mar  2 15:46:38]   == Auto fallthrough, channel
'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL'

I re-register 2 softphones (gerrie001 & gerrie005) :

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
Realtime  
gerrie002/gerrie002(Unspecified)D   N  0
UNREACHABLE Cached RT 
gerrie001/gerrie001192.168.1.105D   N  5060 OK
(11 ms) Cached RT 
gerrie005/gerrie005192.168.1.106D   N  5060 OK
(7 ms)  Cached RT 

The SIP-peer 'gerrie002' is still in the cache ! Don't know where this
is coming from ??

I prune again :

asterisk*CLI> sip prune realtime all
3 peers pruned.
1 users pruned.
[Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
[Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
[Mar  2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11

And again no more peers until I re-register :

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
Realtime 


This realtime thing isn't really working out here... What exactly do I
need to do to clear the cache and thus the old SIP-peers so they can no
longer be used ??

Jonas.
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[asterisk-users] Hide time consuming processed by prompt

[Patrick]

Dear Asterisk users,

I have a simple question, but guess the answer is not that simple :-)

What I want to achieve is to "hide" time consuming processing by a
prompt (load of a customer history), stop the prompt and come
back to the dial plan when the information is available. I'm actually
using AGI script.

How can I do this with asterisk ?

Best regards,
Patrick

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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[Mindaugas Kezys]

Sip reload

 

Regards,

Mindaugas Kezys

 

Kolmisoft UAB 

VoIP Billing Solutions

e-mail:   i...@kolmisoft.com

URL:   http://www.kolmisoft.com

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Tuesday, March 02, 2010 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] rtcachefriends & qualify & sip reload

 

On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: 

 
If you are changing RealTime config in your DB you need to do a sip 
prune realtime either directly from asterisk cli or using AMI. You 
really do not need to do a SIP reload when changing the config of one 
sip extension.

I notice that after a "sip prune realtime all" I also loose all of my realtime 
sip peers. Same result actually as with "sip reload".

I close the softphone of gerrie2 (becomes unspecified)

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status 
Realtime  
gerrie005/gerrie005192.168.1.106D   N  5060 OK (4 ms)  
Cached RT 
gerrie002/gerrie002(Unspecified)D   N  0UNKNOWN
Cached RT 
gerrie001/gerrie001192.168.1.105D   N  5060 OK (11 ms) 
Cached RT

I prune the realtime peers to no longer have gerrie002 in cache :

asterisk*CLI> sip prune realtime all
3 peers pruned.
2 users pruned.
[Mar  2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 
'gerrie001' is now UNREACHABLE!  Last qualify: 91

The realtime peers are all gone :

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status 
Realtime 

Internal call fails :

[Mar  2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 20 - Unknown)
[Mar  2 15:46:38]   == Everyone is busy/congested at this time (1:0/0/1)
[Mar  2 15:46:38]   == Auto fallthrough, channel 'SIP/gerrie001-09f631e0' 
status is 'CHANUNAVAIL'

I re-register 2 softphones (gerrie001 & gerrie005) :

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status 
Realtime  
gerrie002/gerrie002(Unspecified)D   N  0UNREACHABLE 
Cached RT 
gerrie001/gerrie001192.168.1.105D   N  5060 OK (11 ms) 
Cached RT 
gerrie005/gerrie005192.168.1.106D   N  5060 OK (7 ms)  
Cached RT 

The SIP-peer 'gerrie002' is still in the cache ! Don't know where this is 
coming from ??

I prune again :

asterisk*CLI> sip prune realtime all
3 peers pruned.
1 users pruned.
[Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 
'gerrie001' is now UNREACHABLE!  Last qualify: 11
[Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 
'gerrie001' is now UNREACHABLE!  Last qualify: 11
[Mar  2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: Peer 
'gerrie001' is now UNREACHABLE!  Last qualify: 11

And again no more peers until I re-register :

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status 
Realtime 


This realtime thing isn't really working out here... What exactly do I need to 
do to clear the cache and thus the old SIP-peers so they can no longer be used 
??

Jonas. 

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Re: [asterisk-users] Hide time consuming processed by prompt

[Steve Edwards]

On Tue, 2 Mar 2010, Patrick wrote:

> What I want to achieve is to "hide" time consuming processing by a 
> prompt (load of a customer history), stop the prompt and come back to 
> the dial plan when the information is available. I'm actually using AGI 
> script.

Many moons ago I wrote an AGI (written in C) to do credit card 
authorizations. To "hide" the time it took to send the request and receive 
the response, I created another thread that played "Please wait while your 
card is authorized" while the "main" program did the auth. The response 
was almost always received before the end of the prompt so the "customer 
experience" was that the process was instantaneous.

The biggest trick was remembering that you can't execute any AGI commands 
(like 'verbose') in the mainline until the prompt thread completes.

You can execute XXXs of AGIs written in C in the time it takes to load the 
Perl or PHP interpreter.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Echo cancellation on DAHDI

[Vinícius Fontes]

- "Brian"  escreveu:

> On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
> > - "DHAVAL INDRODIYA"  escreveu:
> >
> > > Dear All,
> > >
> > > How can we know the On board supports echo cancellation
> > >
> > > I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V
> (rev
> > > 02) board
> > >
> > > all working fine but sometimes i got echo when user are calling a
> PRI.
> > >
> > > is there any way to know on board echo cancellation .
> > >
> > >
> > > regards
> > >
> > > Dhaval
> >
> > Do you have an echo cancelling module attached to that board? If so,
> all you need is to set echocancel=yes and echocancelwhenbridged=no on
> your chan_dahdi.conf. If you don't... well you should!
> >
> > Anyway, you can turn on the echocancelling via software with
> echocancel=256. I strongly recommend using OSLEC in that case. You'll
> need to patch your DAHDI in order to use it, but it's totally worth
> it.
> >
> On the subject of DAHDI -v- OSLEC.
> 
> I never had any luck getting it to work with DAHDI 2.2.1 despite
> following:
> 
> http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi
> 
> All I ever go was a bad case of the blues :-(
> 
> make[3]: *** No rule to make target
> `/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by
> `/usr/src/dahdi/linux/drivers/dahdi/echo.o'.
> 
> I guess I missed something somewhere???
> 

Get the most recent version of Linux 2.6 kernel. Inside you'll find a directory 
named staging/echo. Copy that entire directory to the drivers/linux directory 
of the DAHDI sources. In the end you gotta have a directory named 
linux/drivers/staging/echo inside your DAHDI sources.

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Re: [asterisk-users] Server response time

[Juan C. Villa]

Gordon,

Thank you very much for the detailed insights! I really appreciate it. I'm 
gonna test drive a server in Germany today. The main reason for choosing a 
server in Germany is COST ($65 vs $200).

Thanks!


-
Juan C. Villa
Computer Engineering
Georgia Institute of Technology
juan...@gatech.edu
(404)441-9653


- Original Message -
From: Gordon Henderson
[mailto:gordon+aster...@drogon.net]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com]
Sent:
Tue, 02 Mar 2010 00:52:51 -0800
Subject: Re: [asterisk-users] Server
response time


> On Mon, 1 Mar 2010, Juan C. Villa wrote:
> 
> > On 2/28/2010 10:21 AM, Gordon Henderson wrote:
> >> On Sun, 28 Feb 2010, Juan C. Villa wrote:
> >>
> >>> Hey Guys,
> >>>
> >>> I am considering leasing a new server in Germany to run my Asterisk
> >>> infrastructure and I was wondering how response time would affect the
> >>> performance of the system. Right now I have a response time of around
> >>> 60-70ms with my server in California. The server in Germany would have a
> >>> response time of around 140ms (both ways). My DID/Termination providers
> >>> are in Canada and the USA, and all my voip boxes are also in the USA.
> >>> Any suggestions or recommendations?
> >>>
> >> Being based in the UK, I'd say why not the UK rather then Germany - we're
> >> closer to the US after-all :)
> >>
> >> However, one thing we don't know: Where are you and your customers based?
> >>
> >> I also find it odd that a lot of people UK based still think they can get
> >> better deals (cheaper&  more b/w) by hosting in the US rather than in the
> >> UK - so I'm curious as to why you'd want to host outside the US...
> >>
> >> But as long as you're not passing media then anywhere you have good
> >> connectivity ought to work - however if you are passing media, then I'd
> be
> >> concerned that someone in California is calling their neighbour and the
> >> data is going all the way to Germany and back again... That really will
> be
> >> noticeable...
> >>
> >
> > In response to Gordon: Hetzner offers the best dedicated server deal I
> > have every seen. I have been a Cari.net client for  over a year now, but
> > I am needing a more powerful server and I don't want to pay $200+ a
> > month for it. Hetzner has a connection to the Level 3 network that
> > recently installed a transoceanic fiber optic link with a lag of less
> > than 40 ms.
> 
> You're not going to get much better than 40ms each way from NY to Europe 
> because as Scotty would say: Ye canny break the laws o' physics! (Actually 
> light in fibre takes 26.1ms according to Wolfram alpha but London to NY 
> has been ~40ms each way since as long as I've been involved with that 
> stuff (mid 90's)
> 
> And most big ISPs in europe now connect to Level3 - e.g. the co-lo I use 
> in deepest darkest england (nowhere near London, although we do have Gb to 
> London) has a ping time like:
> 
>gordon @ unicorn: ping -q -c10 www.nyiix.net
>PING ns3.nyiix.net (209.137.140.21) 56(84) bytes of data.
> 
>--- ns3.nyiix.net ping statistics ---
>10 packets transmitted, 10 received, 0% packet loss, time 9009ms
>rtt min/avg/max/mdev = 78.865/79.947/86.082/2.079 ms
> 
> It goes via L3 and that's probably not the best end-point, but it's close 
> enough, and ~40ms each way.
> 
> > The total lag from Germany to USA (2 way) is around ~110ms (Just tested
> > it today). Who this cause any issues with my VoIP applications? Right
> > now I have two VoIP boxes installed in Switzerland which are connected
> > to my server in California (avg response time = 190ms) and I have no
> > problems at all. What would you guys advice?
> 
> So are you passing data, or just signalling? If data, then why? (Although 
> I guess you're actually terminating to the PSTN in those countries?) But 
> as you already have servers in Switzerland, why can't you use those to run 
> some extended tests, and work it out for yourself?
> 
> Personally, I'd not even think about servers in another country unless I 
> had good reason to - and good "remote hands"/support, etc. and a 
> requirement to plumb in to the local PSTN - either directly or via a local 
> VoIP carrier - and even then, if it's via a local VoIP carrier - why not 
> just connect directly to them from 'home' rather than put a box over 
> there.
> 
> But I if you already have servers in .ch which you indicate you're happy 
> with, then I guess you do have good reason to have them there, so since 
> .de is just up the road from .ch, then if you're happy with the ISP/co-lo 
> then go for it...
> 
> Do make sure the facility has multiple carrier ISPs though - if L3 does go 
> down (and no-ones perfect), you still need a way to get to it - L3 isn't 
> the only backhaul ISP with trans-atlantic links - get the co-lo's AS 
> number and see who they're peering with using the various 'whois' tools, 
> etc.
> 
> Good luck!
> 
> Gordon
> 
> -- 
> _

Re: [asterisk-users] Echo cancellation on DAHDI

[Brian]

On Tue, 2010-03-02 at 12:45 -0300, Vinícius Fontes wrote:
> - "Brian"  escreveu:
> 
> > On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
> > > - "DHAVAL INDRODIYA"  escreveu:
> > >
> > > > Dear All,
> > > >
> > > > How can we know the On board supports echo cancellation
> > > >
> > > > I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V
> > (rev
> > > > 02) board
> > > >
> > > > all working fine but sometimes i got echo when user are calling a
> > PRI.
> > > >
> > > > is there any way to know on board echo cancellation .
> > > >
> > > >
> > > > regards
> > > >
> > > > Dhaval
> > >
> > > Do you have an echo cancelling module attached to that board? If so,
> > all you need is to set echocancel=yes and echocancelwhenbridged=no on
> > your chan_dahdi.conf. If you don't... well you should!
> > >
> > > Anyway, you can turn on the echocancelling via software with
> > echocancel=256. I strongly recommend using OSLEC in that case. You'll
> > need to patch your DAHDI in order to use it, but it's totally worth
> > it.
> > >
> > On the subject of DAHDI -v- OSLEC.
> > 
> > I never had any luck getting it to work with DAHDI 2.2.1 despite
> > following:
> > 
> > http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi
> > 
> > All I ever go was a bad case of the blues :-(
> > 
> > make[3]: *** No rule to make target
> > `/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by
> > `/usr/src/dahdi/linux/drivers/dahdi/echo.o'.
> > 
> > I guess I missed something somewhere???
> > 
> 
> Get the most recent version of Linux 2.6 kernel. Inside you'll find a 
> directory named staging/echo. Copy that entire directory to the drivers/linux 
> directory of the DAHDI sources. In the end you gotta have a directory named 
> linux/drivers/staging/echo inside your DAHDI sources.

I already have those :-(
ls -alh /usr/src/dahdi/dahdi/linux/drivers/staging/echo
drwxr-xr-x 2 root root 4.0K 2010-03-02 14:04 .
drwxr-xr-x 3 root root 4.0K 2010-03-02 14:04 ..
-rw-r--r-- 1 root root 5.7K 2010-03-02 14:04 bit_operations.h
-rw-r--r-- 1 root root  20K 2010-03-02 14:04 echo.c
-rw-r--r-- 1 root root 7.2K 2010-03-02 14:04 echo.h
-rw-r--r-- 1 root root 7.4K 2010-03-02 14:04 fir.h
-rw-r--r-- 1 root root  251 2010-03-02 14:04 Kconfig
-rw-r--r-- 1 root root   29 2010-03-02 14:04 Makefile
-rw-r--r-- 1 root root  14K 2010-03-02 14:04 mmx.h
-rw-r--r-- 1 root root 2.8K 2010-03-02 14:04 oslec.h
-rw-r--r-- 1 root root  367 2010-03-02 14:04 TODO

To be sure I copied them again...
cp
-rf /usr/src/dahdi/linux-2.6.28/drivers/staging/echo/* 
/usr/src/dahdi/dahdi/linux/drivers/staging/echo
(the /dahdi/dahdi is not a typo...)

But still no dice :-(

/usr/src/dahdi/dahdi# make
make -C linux all
make[1]: Entering directory `/usr/src/dahdi/dahdi/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory
`/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware'
make[2]: Leaving directory
`/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware'
make -C /lib/modules/2.6.27-7-server/build
SUBDIRS=/usr/src/dahdi/dahdi/linux/drivers/dahdi
DAHDI_INCLUDE=/usr/src/dahdi/dahdi/linux/include DAHDI_MODULES_EXTRA=" "
HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m
make[2]: Entering directory `/usr/src/linux-headers-2.6.27-7-server'
make[3]: *** No rule to make target
`/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.c', needed by
`/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.o'.  Stop.
make[2]: *** [_module_/usr/src/dahdi/dahdi/linux/drivers/dahdi] Error 2
make[2]: Leaving directory `/usr/src/linux-headers-2.6.27-7-server'
make[1]: *** [modules] Error 2
make[1]: Leaving directory `/usr/src/dahdi/dahdi/linux'
make: *** [all] Error 2

It would be nice to resolve this - but it's probably beyond my
understanding and ability.


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Re: [asterisk-users] Hide time consuming processed by prompt

[Kevin P. Fleming]

Steve Edwards wrote:
> On Tue, 2 Mar 2010, Patrick wrote:
> 
>> What I want to achieve is to "hide" time consuming processing by a 
>> prompt (load of a customer history), stop the prompt and come back to 
>> the dial plan when the information is available. I'm actually using AGI 
>> script.

This sort of thing is easy to do using ExternalIVR instead of AGI.

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

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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[Ishfaq Malik]

jonas kellens wrote:
> On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
>> If you are changing RealTime config in your DB you need to do a sip 
>> prune realtime either directly from asterisk cli or using AMI. You 
>> really do not need to do a SIP reload when changing the config of one 
>> sip extension.
>> 
> I notice that after a "sip prune realtime all" I also loose all of my 
> realtime sip peers. Same result actually as with "sip reload".
>
> I close the softphone of gerrie2 (becomes unspecified)
>
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port 
> Status Realtime 
> gerrie005/gerrie005192.168.1.106D   N  5060 OK 
> (4 ms)  Cached RT
> gerrie002/gerrie002(Unspecified)D   N  0
> UNKNOWNCached RT
> gerrie001/gerrie001192.168.1.105D   N  5060 OK 
> (11 ms) Cached RT
>
> I prune the realtime peers to no longer have gerrie002 in cache :
>
> asterisk*CLI> sip prune realtime all
> 3 peers pruned.
> 2 users pruned.
> [Mar  2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: 
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 91
you're doing the wrong thing!

If you want to get rid of just gerrie002 you need to do

sip prune realtime gerrie002

That will clear only gerrie002 from the realtime cache and leave the 
others alone.
>
> The realtime peers are all gone :
>
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port 
> Status Realtime
>
> Internal call fails :
>
> [Mar  2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable 
> to create channel of type 'SIP' (cause 20 - Unknown)
> [Mar  2 15:46:38]   == Everyone is busy/congested at this time (1:0/0/1)
> [Mar  2 15:46:38]   == Auto fallthrough, channel 
> 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL'
>
> I re-register 2 softphones (gerrie001 & gerrie005) :
>
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port 
> Status Realtime 
> gerrie002/gerrie002(Unspecified)D   N  0
> UNREACHABLE Cached RT
> gerrie001/gerrie001192.168.1.105D   N  5060 OK 
> (11 ms) Cached RT
> gerrie005/gerrie005192.168.1.106D   N  5060 OK 
> (7 ms)  Cached RT
>
> The SIP-peer 'gerrie002' is still in the cache ! Don't know where this 
> is coming from ??
>
> I prune again :
>
> asterisk*CLI> sip prune realtime all
> 3 peers pruned.
> 1 users pruned.
> [Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: 
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
> [Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: 
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
> [Mar  2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer: 
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
>
> And again no more peers until I re-register :
>
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port 
> Status Realtime
>
>
> This realtime thing isn't really working out here... What exactly do I 
> need to do to clear the cache and thus the old SIP-peers so they can 
> no longer be used ??
>
> Jonas. 

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] Echo cancellation on DAHDI

[Gordon Henderson]

On Tue, 2 Mar 2010, Brian wrote:

> It would be nice to resolve this - but it's probably beyond my
> understanding and ability.

Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi 
directory?

Gordon

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[asterisk-users] realtime call peers status

[lore]

Hi all,
I need to check in realtime the calls that my asterisk is menaging:
1) SIP peers status and with who are talking.
2) IAX peers status and with who are talking
3) elapsed talking time

Some one could show me the way to realize that?

Any help are really appreciated

Thanks a lot in advance

-- 
"Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che
non conta un cazzo, 1941 ... sono anche un autore"

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Re: [asterisk-users] Server response time

[Gordon Henderson]

On Tue, 2 Mar 2010, Juan C. Villa wrote:

> Gordon,
>
> Thank you very much for the detailed insights! I really appreciate it. 
> I'm gonna test drive a server in Germany today. The main reason for 
> choosing a server in Germany is COST ($65 vs $200).

I'm very surprised to hear that co-lo's in the US charge that much a month 
for a server. That must be one hellofa server and bandwidth package!

I'd still be wary of sending media over the atlantic and back again though 
- it's a lot on Internet to rely on, even if the ping times are low 
enough.

Gordon

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Re: [asterisk-users] Echo cancellation on DAHDI

[Brian]

On Tue, 2010-03-02 at 15:59 +, Brian wrote:
> On Tue, 2010-03-02 at 12:45 -0300, Vinícius Fontes wrote:
> > - "Brian"  escreveu:
> > 
> > > On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
> > > > - "DHAVAL INDRODIYA"  escreveu:
> > > >
> > > > > Dear All,
> > > > >
> > > > > How can we know the On board supports echo cancellation
> > > > >
> > > > > I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V
> > > (rev
> > > > > 02) board
> > > > >
> > > > > all working fine but sometimes i got echo when user are calling a
> > > PRI.
> > > > >
> > > > > is there any way to know on board echo cancellation .
> > > > >
> > > > >
> > > > > regards
> > > > >
> > > > > Dhaval
> > > >
> > > > Do you have an echo cancelling module attached to that board? If so,
> > > all you need is to set echocancel=yes and echocancelwhenbridged=no on
> > > your chan_dahdi.conf. If you don't... well you should!
> > > >
> > > > Anyway, you can turn on the echocancelling via software with
> > > echocancel=256. I strongly recommend using OSLEC in that case. You'll
> > > need to patch your DAHDI in order to use it, but it's totally worth
> > > it.
> > > >
> > > On the subject of DAHDI -v- OSLEC.
> > > 
> > > I never had any luck getting it to work with DAHDI 2.2.1 despite
> > > following:
> > > 
> > > http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi
> > > 
> > > All I ever go was a bad case of the blues :-(
> > > 
> > > make[3]: *** No rule to make target
> > > `/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by
> > > `/usr/src/dahdi/linux/drivers/dahdi/echo.o'.
> > > 
> > > I guess I missed something somewhere???
> > > 
> > 
> > Get the most recent version of Linux 2.6 kernel. Inside you'll find a 
> > directory named staging/echo. Copy that entire directory to the 
> > drivers/linux directory of the DAHDI sources. In the end you gotta have a 
> > directory named linux/drivers/staging/echo inside your DAHDI sources.
> 
> I already have those :-(
> ls -alh /usr/src/dahdi/dahdi/linux/drivers/staging/echo
> drwxr-xr-x 2 root root 4.0K 2010-03-02 14:04 .
> drwxr-xr-x 3 root root 4.0K 2010-03-02 14:04 ..
> -rw-r--r-- 1 root root 5.7K 2010-03-02 14:04 bit_operations.h
> -rw-r--r-- 1 root root  20K 2010-03-02 14:04 echo.c
> -rw-r--r-- 1 root root 7.2K 2010-03-02 14:04 echo.h
> -rw-r--r-- 1 root root 7.4K 2010-03-02 14:04 fir.h
> -rw-r--r-- 1 root root  251 2010-03-02 14:04 Kconfig
> -rw-r--r-- 1 root root   29 2010-03-02 14:04 Makefile
> -rw-r--r-- 1 root root  14K 2010-03-02 14:04 mmx.h
> -rw-r--r-- 1 root root 2.8K 2010-03-02 14:04 oslec.h
> -rw-r--r-- 1 root root  367 2010-03-02 14:04 TODO
> 
> To be sure I copied them again...
> cp
> -rf /usr/src/dahdi/linux-2.6.28/drivers/staging/echo/* 
> /usr/src/dahdi/dahdi/linux/drivers/staging/echo
> (the /dahdi/dahdi is not a typo...)
> 
> But still no dice :-(
> 
> /usr/src/dahdi/dahdi# make
> make -C linux all
> make[1]: Entering directory `/usr/src/dahdi/dahdi/linux'
> make -C drivers/dahdi/firmware firmware-loaders
> make[2]: Entering directory
> `/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware'
> make[2]: Leaving directory
> `/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware'
> make -C /lib/modules/2.6.27-7-server/build
> SUBDIRS=/usr/src/dahdi/dahdi/linux/drivers/dahdi
> DAHDI_INCLUDE=/usr/src/dahdi/dahdi/linux/include DAHDI_MODULES_EXTRA=" "
> HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m
> make[2]: Entering directory `/usr/src/linux-headers-2.6.27-7-server'
> make[3]: *** No rule to make target
> `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.c', needed by
> `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.o'.  Stop.
> make[2]: *** [_module_/usr/src/dahdi/dahdi/linux/drivers/dahdi] Error 2
> make[2]: Leaving directory `/usr/src/linux-headers-2.6.27-7-server'
> make[1]: *** [modules] Error 2
> make[1]: Leaving directory `/usr/src/dahdi/dahdi/linux'
> make: *** [all] Error 2
> 
> It would be nice to resolve this - but it's probably beyond my
> understanding and ability.

Actually - looking at that
make[3]: *** No rule to make target
`/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.c', needed by
`/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.o'.  Stop.

There is no echo.c in /usr/src/dahdi/dahdi/linux/drivers/dahdi/ - 
That file is in /usr/src/dahdi/dahdi/linux/drivers/staging/echo/

I've followed this with care:
http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi

So I'm stumped...




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Re: [asterisk-users] realtime call peers status

[Ishfaq Malik]

lore wrote:
> Hi all,
> I need to check in realtime the calls that my asterisk is menaging:
> 1) SIP peers status and with who are talking.
> 2) IAX peers status and with who are talking
> 3) elapsed talking time
>
> Some one could show me the way to realize that?
>
> Any help are really appreciated
>
> Thanks a lot in advance
>
>   

 From asterisk cli

core show channels
core show channel 

If you need to put it into a pretty front end you can use the AMI

Ish
-- 
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Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject - Bayesian Filter detected spam

[DLeese]

Hi,

I have tried the new revision (769) with Asterisk SVN-trunk-r240667M (~ 1.6.2) 
and it compiles without warnings or errors (see attached make output). It also 
seems to work flawlessly. I can make and receive calls from/to the PSTN with 
the Fritz card PCI via the BRI and route them to my voip Telephones.

I will also try faxing, but not today. 

Many thanks for the great work!

Sincerly 

Daniel Leese


---

srvpbx:/usr/src/chan-capi-trunk# make
./create_config.sh "/usr/include"
Checking Asterisk version... SVN-trunk-r240667M
 * assuming Asterisk version 1.6
Using Asterisk 1.6 API
 * found new 'ast_dsp_set_digitmode' function
 * found new union data in ast_frame structure
 * found new union subclass in ast_frame structure
 * found ast_channel_release function
 * found new ast_devstate2str function
 * found requestor in ast_request
 * found format_t in ast_request
 * found const char in ast_register_application
 * found linkedid in ast_channel_alloc
 * found format_t in frame_defs
 * found rtp_engine.h
config.h complete.

 [CC] chan_capi.c -> chan_capi.o
 [CC] chan_capi_utils.c -> chan_capi_utils.o
 [CC] chan_capi_rtp.c -> chan_capi_rtp.o
 [CC] chan_capi_command.c -> chan_capi_command.o
 [CC] xlaw.c -> xlaw.o
 [CC] dlist.c -> dlist.o
 [CC] chan_capi_qsig_core.c -> chan_capi_qsig_core.o
 [CC] chan_capi_qsig_ecma.c -> chan_capi_qsig_ecma.o
 [CC] chan_capi_qsig_asn197ade.c -> chan_capi_qsig_asn197ade.o
 [CC] chan_capi_qsig_asn197no.c -> chan_capi_qsig_asn197no.o
 [CC] chan_capi_supplementary.c -> chan_capi_supplementary.o
 [CC] chan_capi_chat.c -> chan_capi_chat.o
 [CC] libcapi20/convert.c -> libcapi20/convert.o
 [CC] libcapi20/capi20.c -> libcapi20/capi20.o
 [CC] libcapi20/capifunc.c -> libcapi20/capifunc.o
 [LD] chan_capi.so (chan_capi.o chan_capi_utils.o chan_capi_rtp.o 
chan_capi_command.o xlaw.o dlist.o chan_capi_qsig_core.o chan_capi_qsig_ecma.o 
chan_capi_qsig_asn197ade.o chan_capi_qsig_asn197no.o chan_capi_supplementary.o 
chan_capi_chat.o libcapi20/convert.o libcapi20/capi20.o libcapi20/capifunc.o)
srvpbx:/usr/src/chan-capi-trunk#

> chan_capi trunk should be compilable now with current asterisk trunk.
> 
> Armin
> 
> 
> On Mon, 1 Mar 2010, Armin Schindler wrote:
> > Hi,
> >
> > it seems that the asterisk API here was changed again and chan_capi 
> > must be adapted to this. I will have a look.
> >
> > Armin
> >
> > On Mon, 1 Mar 2010, dle...@lstelcom.com wrote:
> >> Hi again!
> >> 
> >>> I have excellent success with the tiny "fcpci" and 
> chan_capi, which 
> >>> is also working great with capi4hylafax. See
> >>> net-dialup/fcpci-0.1-r1 in gentoo (should not be difficult to use 
> >>> this on other distros, but I have never done so). Do not confuse 
> >>> this with the "fritzcapi"!
> >> 
> >> I managed to install fcpci and it seems to run fine 
> (capiinfo output). 
> >> Unfortunately i cant compile chan_capi against my Asterisk 
> trunk r240716. 
> >> Neither the trunk/head nor the 1.1.4 Version compiles. All 
> fail with 
> >> the following output:
> >> 
> >> srvpbx:/usr/src/chan-capi-HEAD# make
> >> [CC] chan_capi.c -> chan_capi.o
> >> In file included from chan_capi.c:32:
> >> chan_capi.h:34:26: error: asterisk/rtp.h: Datei oder Verzeichnis 
> >> nicht gefunden
> >> chan_capi.c: In function âlocal_queue_frameâ:
> >> chan_capi.c:803: error: invalid operands to binary == (have âunion 
> >> â and âintâ)
> >> chan_capi.c: In function âinterface_cleanupâ:
> >> chan_capi.c:1071: warning: implicit declaration of function 
> >> âast_rtp_destroyâ
> >> chan_capi.c: In function âsend_progressâ:
> >> chan_capi.c:1165: error: incompatible types in assignment
> >> chan_capi.c: In function âclear_channel_fax_loopâ:
> >> chan_capi.c:2884: error: invalid operands to binary == 
> (have âunion 
> >> â and âintâ)
> >> chan_capi.c: In function âcapidev_handle_did_digitsâ:
> >> chan_capi.c:3548: error: incompatible types in assignment
> >> chan_capi.c: In function âcapi_queue_cause_controlâ:
> >> chan_capi.c:3564: warning: missing braces around initializer
> >> chan_capi.c:3564: warning: (near initialization for âfr.subclassâ)
> >> chan_capi.c:3569: error: incompatible types in assignment
> >> chan_capi.c:3573: error: incompatible types in assignment
> >> chan_capi.c: In function âcapidev_handle_info_indicationâ:
> >> chan_capi.c:3876: error: incompatible types in assignment
> >> chan_capi.c:3886: error: incompatible types in assignment
> >> chan_capi.c: In function âhandle_facility_indication_dtmfâ:
> >> chan_capi.c:4138: error: incompatible types in assignment
> >> chan_capi.c:4149: error: incompatible types in assignment
> >> chan_capi.c: In function âcapidev_handle_data_b3_indicationâ:
> >> chan_capi.c:4292: error: incompatible types in assignment
> >> chan_capi.c:4294: error: incompatible types in assignment
> >> chan_capi.c: In function âcapi_signal_answerâ:
> >> chan_capi.c:4316: warning: missing braces around initializer
> >> chan_capi.c:4316: warning: (near initialization for âfr.subclassâ)
> >> chan_c

Re: [asterisk-users] Echo cancellation on DAHDI

[Brian]

On Tue, 2010-03-02 at 16:27 +, Gordon Henderson wrote:
> On Tue, 2 Mar 2010, Brian wrote:
> 
> > It would be nice to resolve this - but it's probably beyond my
> > understanding and ability.
> 
> Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi 
> directory?
> 
> Gordon
> 
Ah Gordon! Thank God you are here!

No my friend, I did not. I was blindly following this

http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi

But looking at Kbuild all it has in it is...


obj-m += echo.o


So I don't have two lines to uncomment ??? Methinks something ain't
right here. Colombo would be proud of me.

It appears there are issues with make -v- the path of echo.c + echo.o
from my limited comprehension of such matters.

Perhaps I'll whirl it again, this time unpacking to /usr/src/dahdi
rather than /usr/src/dahdi/dahdi. I had adjusted the paths but just in
case.. .. .. ..


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[asterisk-users] ARI problem with monitor

[Luis campo]

Dear Sirs,

I
installed AsteriskNOW 1.5 and the CDR is working with mysql as unique
ID, when you use call recording these files are stored in / var / spool
/ asterisk / monitor. but wanting to see through the ARI monitor leaves the 
column blank.

What could be the problem.

Luis
  
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Re: [asterisk-users] Echo cancellation on DAHDI

[Carlos Chavez]

On Tue, 2010-03-02 at 15:44 +0530, DHAVAL INDRODIYA wrote:
> Dear All,
> 
> How can we know the On board supports echo cancellation 
> 
> I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
> 02) board 
> 
> all working fine but sometimes i got echo when user are calling a PRI.
> 
> is there any way to know on board echo cancellation .
> 
> 
Check "dmesg" on your system for messages like:

VPM400: Support Enabled/Disabled
VPM450: Support Enabled/Disabled

That should tell you if the hardware echo cancellation is working or
not.  The TE410P does not have hardware echo cancellation the model was
TE411P.  If you can open the server you should be able to see if the
card has a daughter board installed which is the echo module.

> regards
> 
> Dhaval
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Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] realtime call peers status

[lore]

Hi,
thanks a lot for the reply,
yes I would like to put data in a web interface (maybe php made better
if already done :) ).
I'm reading something about dymanic realtime: could be ok for my needs?
Or is better spent my time on this docs :
http://www.voip-info.org/wiki/view/Asterisk+manager+API ?



2010/3/2 Ishfaq Malik :
> lore wrote:
>> Hi all,
>> I need to check in realtime the calls that my asterisk is menaging:
>> 1) SIP peers status and with who are talking.
>> 2) IAX peers status and with who are talking
>> 3) elapsed talking time
>>
>> Some one could show me the way to realize that?
>>
>> Any help are really appreciated
>>
>> Thanks a lot in advance
>>
>>
>
>  From asterisk cli
>
> core show channels
> core show channel 
>
> If you need to put it into a pretty front end you can use the AMI
>
> Ish
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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Re: [asterisk-users] rtcachefriends & qualify & sip reload

[Carlos Chavez]

On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote:
> On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote: 
> > If you are changing RealTime config in your DB you need to do a sip 
> > prune realtime either directly from asterisk cli or using AMI. You 
> > really do not need to do a SIP reload when changing the config of one 
> > sip extension.
> I notice that after a "sip prune realtime all" I also loose all of my
> realtime sip peers. Same result actually as with "sip reload".
> 
> I close the softphone of gerrie2 (becomes unspecified)
> 
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port Status
> Realtime  
> gerrie005/gerrie005192.168.1.106D   N  5060 OK
> (4 ms)  Cached RT 
> gerrie002/gerrie002(Unspecified)D   N  0
> UNKNOWNCached RT 
> gerrie001/gerrie001192.168.1.105D   N  5060 OK
> (11 ms) Cached RT
> 
> I prune the realtime peers to no longer have gerrie002 in cache :
> 
> asterisk*CLI> sip prune realtime all
> 3 peers pruned.
> 2 users pruned.
> [Mar  2 15:42:19] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 91
> 
> The realtime peers are all gone :
> 
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port Status
> Realtime 
> 
> Internal call fails :
> 
> [Mar  2 15:46:38] WARNING[558]: app_dial.c:1272 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 20 - Unknown)
> [Mar  2 15:46:38]   == Everyone is busy/congested at this time
> (1:0/0/1)
> [Mar  2 15:46:38]   == Auto fallthrough, channel
> 'SIP/gerrie001-09f631e0' status is 'CHANUNAVAIL'
> 
> I re-register 2 softphones (gerrie001 & gerrie005) :
> 
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port Status
> Realtime  
> gerrie002/gerrie002(Unspecified)D   N  0
> UNREACHABLE Cached RT 
> gerrie001/gerrie001192.168.1.105D   N  5060 OK
> (11 ms) Cached RT 
> gerrie005/gerrie005192.168.1.106D   N  5060 OK
> (7 ms)  Cached RT 
> 
> The SIP-peer 'gerrie002' is still in the cache ! Don't know where this
> is coming from ??
> 
> I prune again :
> 
> asterisk*CLI> sip prune realtime all
> 3 peers pruned.
> 1 users pruned.
> [Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
> [Mar  2 15:51:57] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
> [Mar  2 15:52:01] NOTICE[32498]: chan_sip.c:16612 sip_poke_noanswer:
> Peer 'gerrie001' is now UNREACHABLE!  Last qualify: 11
> 
> And again no more peers until I re-register :
> 
> asterisk*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port Status
> Realtime 
> 
> 
> This realtime thing isn't really working out here... What exactly do I
> need to do to clear the cache and thus the old SIP-peers so they can
> no longer be used ??
> 

Do not prune all peers, only the peer you wish to reload or eliminate!
Do "sip prune realtime peer peername".  That way you do not lose all the
other registrations.  I really do not see this as a problem as the
phones will usually re register quickly or if the user dials any number.

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Re: [asterisk-users] asterisk-users Digest, Vol 68, Issue 4

[Luis Silva]

Hi Philipe

>
>Hi!

>> I started to have problems with sip trunks, using more than one trunk
>> (and sometimes using only one) the sip module seems to freeze... My local
>> extensions lost registration and also the trunks. The only way that I
>> can restart the sip is removing the trunks...

>Have seen this also on different 1.4 versions (1.4.18), created a bug
>report and was told this to be a DNS issue. I do not think it is, but
>never got this one fully solved.

>https://issues.asterisk.org/view.php?id=15139
>https://issues.asterisk.org/view.php?id=15052

>'This bug is surely a DNS related problem, if you look over /main/dns.c
>you'll find:
>"Asterisk DNS is synchronus at this time. This means that if your DNS
>does not work properly, Asterisk might not start properly or a channel
>may lock."
>Try to change nameservers in /etc/resolv.conf.'

DNS has a thing that I suspected, I had some issues with that in the pass...
Where I don't think that is the problem, I tried with two different
network's, lan and isp...

>What distribution & kernel are you using?
Centos 5.4, with a recent yum update, kernel 2.6.18-164.11.1.el5 

>> If I make sip reload or restart asterisk the sip module takes many many
time
>> before starting.

>That I didn't notice, had to do a reboot (!).
I also did that several times, but waiting the sip restarts but after a
while the problem restarts also...


>Philipp

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Re: [asterisk-users] Echo cancellation on DAHDI

[Brian]

On Tue, 2010-03-02 at 16:49 +, Brian wrote:
> On Tue, 2010-03-02 at 16:27 +, Gordon Henderson wrote:
> > On Tue, 2 Mar 2010, Brian wrote:
> > 
> > > It would be nice to resolve this - but it's probably beyond my
> > > understanding and ability.
> > 
> > Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi 
> > directory?
> > 
> > Gordon
> > 
> Ah Gordon! Thank God you are here!
> 
> No my friend, I did not. I was blindly following this
> 
> http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi
> 
> But looking at Kbuild all it has in it is...
> 
> 
> obj-m += echo.o
> 
> 
> So I don't have two lines to uncomment ??? Methinks something ain't
> right here. Colombo would be proud of me.
> 
> It appears there are issues with make -v- the path of echo.c + echo.o
> from my limited comprehension of such matters.
> 
> Perhaps I'll whirl it again, this time unpacking to /usr/src/dahdi
> rather than /usr/src/dahdi/dahdi. I had adjusted the paths but just in
> case.. .. .. ..
> 
> 
My issue was nothing more complex than having downloaded the full dahdi
package. The result is it unpacks to:

/usr/src/dahdi/linux/drivers/staging -
not /usr/src/dahdi/drivers/staging.

Fix was nothing more simple than moving the contents
of /usr/src/dahdi/linux/ to /usr/src/dahdi/ and the 'howto' worked
pretty much like a charm :-)


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Re: [asterisk-users] Server response time

[Juan C. Villa]

On 3/2/2010 12:29 PM, Gordon Henderson wrote:
> On Tue, 2 Mar 2010, Juan C. Villa wrote:
>
>
>> Gordon,
>>
>> Thank you very much for the detailed insights! I really appreciate it.
>> I'm gonna test drive a server in Germany today. The main reason for
>> choosing a server in Germany is COST ($65 vs $200).
>>  
> I'm very surprised to hear that co-lo's in the US charge that much a month
> for a server. That must be one hellofa server and bandwidth package!
>
> I'd still be wary of sending media over the atlantic and back again though
> - it's a lot on Internet to rely on, even if the ping times are low
> enough.
>
> Gordon
>
>

I was very surprised with the price as well. The best package I could 
find in the US was with a company called Server Beach (a Peer1 company). 
And in Germany I found Hetzner. Check out these specs and prices:

Server Beach Specs:
1 x QuadCore
1 x 250gb drive
2 gb RAM
10mbps port (2T capped)
For: $99/mo ($0 setup)

Hetzner Specs:
1 x Intel Core i7 920 (Quad Core + HT)
2 x 750gb drive (software raid 1)
8 gb RAM
100mbps port (2T capped, 10mbps after the first 2T)
For: $65/mo ($200 setup)


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[asterisk-users] MWI and 1.6.1

[Dave Poirier]

We are having an issue with Asterisk 1.6.1 and the MWI turning on when a
user doesn't have voicemail. We see random MWI lights come on and the phone
indicates a random number of messages (its been anywhere from 1-14) when a
server reload is done.

I just checked one user, they have no messages old or new and the phone
(Polycom IP330) indicates that they have 2 messages. The user will check for
messages, the system will tell them that they have none and the light goes
out.

I know that starting in 1.6 Asterisk moved from a polling system to an event
based system but it's unclear to me what is causing these events to be
generated. Anyone else experience this? Any tips, suggestions?

Thanks,
Dave
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[asterisk-users] Asterisk and cellphone/GSM voicemailbox

[jonas kellens]

Does Asterisk know when it hits a voicemailbox ?

When calling to a cell-phone or GSM, after some rings and no pickup you
arrive at a voicemailbox.

If Asterisk does not know it's a voicemailbox that has answered the
call, the voicemailbox will contain 60minutes of 'silence'. This is very
expensive 'silence'.

How to avoid this ?

Jonas
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Re: [asterisk-users] Asterisk and cellphone/GSM voicemailbox

[Fred Posner]

On Mar 2, 2010, at 2:37 PM, jonas kellens wrote:

> Does Asterisk know when it hits a voicemailbox ?
> 
> When calling to a cell-phone or GSM, after some rings and no pickup you 
> arrive at a voicemailbox.
> 
> If Asterisk does not know it's a voicemailbox that has answered the call, the 
> voicemailbox will contain 60minutes of 'silence'. This is very expensive 
> 'silence'.
> 
> How to avoid this ?
> 
> Jonas

You can avoid this is several ways... one of the ways I like best is to dial 
with a macro that then requires the recipient to press 1 or some dtmf 
confirmation to accept the call. Very good at avoiding voicemail, cell phone 
service messages, etc.

---fred
http://qxork.com


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[asterisk-users] FW: ARI problem with monitor

[Luis campo]

Please need help!!.


thx

From: lcr_2...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: ARI problem with monitor
Date: Tue, 2 Mar 2010 16:54:51 +








Dear Sirs,

I
installed AsteriskNOW 1.5 and the CDR is working with mysql as unique
ID, when you use call recording these files are stored in / var / spool
/ asterisk / monitor. but wanting to see through the ARI monitor leaves the 
column blank.

What could be the problem.

Luis
  
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Re: [asterisk-users] dialplan reload: not working with large dialplans

[Warren Selby]

On Tuesday, March 2, 2010, Andreas Brodmann  wrote:
> Hi Tzafrir,
>
> yes, I will have to 'anonymize' the dialplan, is this list the right place 
> though?
>
> -Andreas
>
>

How big is your dialplan?  How many lines / file size, etc. Are you
using ael or lua or just the original .conf file?

Thanks,
--Warren Selby
http://www.SelbyTech.com

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[asterisk-users] Uverse, Asterisk and SIP

[sean darcy]

I've just got Uverse installed. I had dsl, but ATT insisted I couldn't 
keep my old dsl, but had to switch to Uverse internet - vdsl.

My setup:

linux box as router : 10.10.11.252

asterisk box: 10.10.11.180

10.10.11.252 is multihomed and connected to the Uverse Residential 
Gateway. I've set it up as DMZplus, and it shows the public ip address 
as eth1. I can ssh into the linux box from outside.

sip worked fine with dsl. I used teliax, junction and direct sip to the 
asterisk box in the office. I can ssh from 10.10.11.180 to the office.

But not now. The asterisk box sends out sip messages, but nothing comes 
in. In the office asterisk box, I don't see the sip messages come in.

Is anybody using sip behind a Uverse RG? Care to share the magic?

sean


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Re: [asterisk-users] Uverse, Asterisk and SIP

[Fred Posner]

On Mar 2, 2010, at 6:27 PM, sean darcy wrote:

> I've just got Uverse installed. I had dsl, but ATT insisted I couldn't 
> keep my old dsl, but had to switch to Uverse internet - vdsl.
> 
> My setup:
> 
> linux box as router : 10.10.11.252
> 
> asterisk box: 10.10.11.180
> 
> 10.10.11.252 is multihomed and connected to the Uverse Residential 
> Gateway. I've set it up as DMZplus, and it shows the public ip address 
> as eth1. I can ssh into the linux box from outside.
> 
> sip worked fine with dsl. I used teliax, junction and direct sip to the 
> asterisk box in the office. I can ssh from 10.10.11.180 to the office.
> 
> But not now. The asterisk box sends out sip messages, but nothing comes 
> in. In the office asterisk box, I don't see the sip messages come in.
> 
> Is anybody using sip behind a Uverse RG? Care to share the magic?
> 
> sean

Sean,

I had att u-verse up until a week ago and loved it. Ran Asterisk behind it with 
great success. (I only left u-verse because of a physical move).

Anyway, by default the u-verse router simply will block upd like noone's 
business. Make sure you have a firewall and then tell the u-verse router to 
open everything to that firewall (and proceed like you did on dsl). If you 
change the mac of your firewall, you'll need to reauth it again.

---fred
http://qxork.com


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Re: [asterisk-users] dialplan reload: not working with large dialplans

[Andreas Brodmann]

Hi Warren,

the dialplan currently holds 1792 lines. It's a plain old .conf file.

-Andreas


2010/3/2 Warren Selby 

> On Tuesday, March 2, 2010, Andreas Brodmann 
> wrote:
> > Hi Tzafrir,
> >
> > yes, I will have to 'anonymize' the dialplan, is this list the right
> place though?
> >
> > -Andreas
> >
> >
>
> How big is your dialplan?  How many lines / file size, etc. Are you
> using ael or lua or just the original .conf file?
>
> Thanks,
> --Warren Selby
> http://www.SelbyTech.com
>
> --
> Thanks,
> --Warren Selby
> http://www.selbytech.com
>
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[asterisk-users] Dial timeout problem with OpenVox A1200P Card / FXS module

[Fábio da Silva Cunha]

   Hello all!

   I having some trouble with a OpenVox A1200P card equiped with 5 FXO 
and 7 FXS ports, all ok with FXO ports, but the FXS ones are having 
some strange problem:

   With a telephone connected to any FXS port, when i dial some 
extension number on this phone, i receive a busy signal.

   My platform is Linux Ubuntu 9.10, Asterisk 1.6 and 
openvox_dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz from OpenVox.

   Any idea?

   Thanks in advance!

   Best Regards,

Fábio da Silva Cunha
f...@vetorial.net
+55 53 8403 4217







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Re: [asterisk-users] Uverse, Asterisk and SIP

[sean darcy]

Fred Posner wrote:
> On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
> 
>> I've just got Uverse installed. I had dsl, but ATT insisted I couldn't 
>> keep my old dsl, but had to switch to Uverse internet - vdsl.
>>
>> My setup:
>>
>> linux box as router : 10.10.11.252
>>
>> asterisk box: 10.10.11.180
>>
>> 10.10.11.252 is multihomed and connected to the Uverse Residential 
>> Gateway. I've set it up as DMZplus, and it shows the public ip address 
>> as eth1. I can ssh into the linux box from outside.
>>
>> sip worked fine with dsl. I used teliax, junction and direct sip to the 
>> asterisk box in the office. I can ssh from 10.10.11.180 to the office.
>>
>> But not now. The asterisk box sends out sip messages, but nothing comes 
>> in. In the office asterisk box, I don't see the sip messages come in.
>>
>> Is anybody using sip behind a Uverse RG? Care to share the magic?
>>
>> sean
> 
> Sean,
> 
> I had att u-verse up until a week ago and loved it. Ran Asterisk behind it 
> with great success. (I only left u-verse because of a physical move).
> 
> Anyway, by default the u-verse router simply will block upd like noone's 
> business. Make sure you have a firewall and then tell the u-verse router to 
> open everything to that firewall (and proceed like you did on dsl). If you 
> change the mac of your firewall, you'll need to reauth it again.
> 
> ---fred
> http://qxork.com
> 
> 

Well, I think I did that by setting the linux box to DMZplus:


View Firewall Summary->View Firewall Details

Current Settings: Custom
Device AllowedApps AppType Protocol PortNumber(s) PublicIP
76.xxx.yyy.zzz  All -   (all)   (all)   76.xxx.yyy.zzz

and

Edit Advanced Firewall Settings

unchecked all the Security Settings, and unchecked all the Attack Detection.

Anything else?

sean



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Re: [asterisk-users] Uverse, Asterisk and SIP

[Warren Selby]

You need to set your firewall public ip to dhcp in order for Uverse  
dmz to work.



Thanks,
--Warren Selby

On Mar 2, 2010, at 8:53 PM, sean darcy  wrote:

> Fred Posner wrote:
>> On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
>>
>>> I've just got Uverse installed. I had dsl, but ATT insisted I  
>>> couldn't
>>> keep my old dsl, but had to switch to Uverse internet - vdsl.
>>>
>>> My setup:
>>>
>>> linux box as router : 10.10.11.252
>>>
>>> asterisk box: 10.10.11.180
>>>
>>> 10.10.11.252 is multihomed and connected to the Uverse Residential
>>> Gateway. I've set it up as DMZplus, and it shows the public ip  
>>> address
>>> as eth1. I can ssh into the linux box from outside.
>>>
>>> sip worked fine with dsl. I used teliax, junction and direct sip  
>>> to the
>>> asterisk box in the office. I can ssh from 10.10.11.180 to the  
>>> office.
>>>
>>> But not now. The asterisk box sends out sip messages, but nothing  
>>> comes
>>> in. In the office asterisk box, I don't see the sip messages come  
>>> in.
>>>
>>> Is anybody using sip behind a Uverse RG? Care to share the magic?
>>>
>>> sean
>>
>> Sean,
>>
>> I had att u-verse up until a week ago and loved it. Ran Asterisk  
>> behind it with great success. (I only left u-verse because of a  
>> physical move).
>>
>> Anyway, by default the u-verse router simply will block upd like  
>> noone's business. Make sure you have a firewall and then tell the u- 
>> verse router to open everything to that firewall (and proceed like  
>> you did on dsl). If you change the mac of your firewall, you'll  
>> need to reauth it again.
>>
>> ---fred
>> http://qxork.com
>>
>>
>
> Well, I think I did that by setting the linux box to DMZplus:
>
>
> View Firewall Summary->View Firewall Details
>
> Current Settings: Custom
> Device   AllowedApps AppType Protocol PortNumber(s) PublicIP
> 76.xxx.yyy.zzzAll-(all)(all)76.xxx.yyy.zzz
>
> and
>
> Edit Advanced Firewall Settings
>
> unchecked all the Security Settings, and unchecked all the Attack  
> Detection.
>
> Anything else?
>
> sean
>
>
>
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[asterisk-users] asterisk-users] how to create a dummy call

[Pham Quy]

Hi all,

What i'm going to do is that enable caller sing while playing a
background music. My approach is using Monitor and Meetme app.
Caller make a call to asterisk, asterisk join caller in to a voice
conference and create a dummy caller which will play music. Monitor app
record both music and singer's voice. 

But i dont know how to create a dummy caller or throw a dummy call in
order to do above task.

Any idea or comment is appreciated.

Thanks
Quyps


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[asterisk-users] how to play background music during record

[Pham Quy]

Hi all, 

The question has already asked here, 

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg98176.html

but it's been two years since then, so is there any better solution with
latest release version?

Quyps


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Re: [asterisk-users] Echo cancellation on DAHDI

[DHAVAL INDRODIYA]

Hi,

Carlos

I checked dmesg on my server and i found following message

what is meaning for this ? i cant understand

VPM400: Not Present
VPM450: echo cancellation for 128 channels
VPM450: hardware DTMF disabled.
VPM450: Present and operational servicing 4 span(s)

regards
Dhaval
On Tue, Mar 2, 2010 at 10:25 PM, Carlos Chavez wrote:

> On Tue, 2010-03-02 at 15:44 +0530, DHAVAL INDRODIYA wrote:
> > Dear All,
> >
> > How can we know the On board supports echo cancellation
> >
> > I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
> > 02) board
> >
> > all working fine but sometimes i got echo when user are calling a PRI.
> >
> > is there any way to know on board echo cancellation .
> >
> >
> Check "dmesg" on your system for messages like:
>
> VPM400: Support Enabled/Disabled
> VPM450: Support Enabled/Disabled
>
>That should tell you if the hardware echo cancellation is working or
> not.  The TE410P does not have hardware echo cancellation the model was
> TE411P.  If you can open the server you should be able to see if the
> card has a daughter board installed which is the echo module.
>
> > regards
> >
> > Dhaval
> > --
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>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
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[asterisk-users] dahdi and oslec

[Chandrakant Solanki]

Hi All,

I have followed below steps to enable echo cancellation.

# cd /usr/src
# wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2
# tar xjf linux-2.6.28.tar.bz2
# tar zxvf dahdi-linux-2.1.0.4.tar.gz
# ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi
# mkdir /usr/src/dahdi/drivers/staging
# cp -fR /usr/src/linux-2.6.28/drivers/staging/echo
/usr/src/dahdi/drivers/staging
# sed -i "s|#obj-m += dahdi_echocan_oslec.o|obj-m += dahdi_echocan_oslec.o|"
/usr/src/dahdi/drivers/dahdi/Kbuild
# sed -i "s|#obj-m += ../staging/echo/|obj-m += ../staging/echo/|"
/usr/src/dahdi/drivers/dahdi/Kbuild
# echo 'obj-m += echo.o' > /usr/src/dahdi/drivers/staging/echo/Kbuild
# cd /usr/src/dahdi
# make
# make install
# cd /usr/src
# tar zxvf dahdi-tools-2.1.0.2.tar.gz
# cd /usr/src/dahdi-tools-2.1.0.2
# ./configure
# make
# make install

# wget http://www.rowetel.com/ucasterisk/downloads/oslec-0.2.tar.gz
# tar xvzf oslec-0.2.tar.gz
# cd oslec-0.2
# make
# insmod kernel/oslec.ko

when i restart /etc/init.d/dahdi service it gives me following error in
/var/log/message

Mar  3 11:06:37 server1 kernel: echo: exports duplicate symbol oslec_hpf_tx
(owned by oslec)
Mar  3 11:06:37 server1 modprobe: WARNING: Error inserting echo
(/lib/modules/2.6.18-92.1.22.el5/staging/echo/echo.ko): Invalid module
format
Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
oslec_create
Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
oslec_update
Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
oslec_free
Mar  3 11:06:37 server1 modprobe: FATAL: Error inserting dahdi_echocan_oslec
(/lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_echocan_oslec.ko): Unknown
symbol in module, or unknown parameter (see dmesg)

# cat /etc/dahdi/system.conf

loadzone= in
defaultzone = in

span=1,1,7,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
echocanceller=oslec,1-15,17-31

Is there anything missing or i am going wrong..

Help me out.

Thanks in advance...



-- 
Regards,

Chandrakant Solanki
-- 
_
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Re: [asterisk-users] dahdi and oslec

[wins mallow]

On Wed, 2010-03-03 at 11:31 +0530, Chandrakant Solanki wrote:
> Hi All,
> 
> I have followed below steps to enable echo cancellation.
> 
> # cd /usr/src
> # wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2
> # tar xjf linux-2.6.28.tar.bz2
> # tar zxvf dahdi-linux-2.1.0.4.tar.gz
> # ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi
> # mkdir /usr/src/dahdi/drivers/staging
> # cp
> -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging
> # sed -i "s|#obj-m += dahdi_echocan_oslec.o|obj-m +=
> dahdi_echocan_oslec.o|" /usr/src/dahdi/drivers/dahdi/Kbuild
> # sed -i "s|#obj-m += ../staging/echo/|obj-m
> += ../staging/echo/|" /usr/src/dahdi/drivers/dahdi/Kbuild
> # echo 'obj-m += echo.o' > /usr/src/dahdi/drivers/staging/echo/Kbuild
> # cd /usr/src/dahdi
> # make
> # make install
> # cd /usr/src
> # tar zxvf dahdi-tools-2.1.0.2.tar.gz
> # cd /usr/src/dahdi-tools-2.1.0.2
> # ./configure
> # make
> # make install
> 
> # wget http://www.rowetel.com/ucasterisk/downloads/oslec-0.2.tar.gz
> # tar xvzf oslec-0.2.tar.gz
> # cd oslec-0.2
> # make
> # insmod kernel/oslec.ko
> 
> when i restart /etc/init.d/dahdi service it gives me following error
> in /var/log/message
> 
> Mar  3 11:06:37 server1 kernel: echo: exports duplicate symbol
> oslec_hpf_tx (owned by oslec)
> Mar  3 11:06:37 server1 modprobe: WARNING: Error inserting echo
> (/lib/modules/2.6.18-92.1.22.el5/staging/echo/echo.ko): Invalid module
> format 
> Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
> oslec_create
> Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
> oslec_update
> Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
> oslec_free
> Mar  3 11:06:37 server1 modprobe: FATAL: Error inserting
> dahdi_echocan_oslec
> (/lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_echocan_oslec.ko):
> Unknown symbol in module, or unknown parameter (see dmesg) 
> 
> # cat /etc/dahdi/system.conf 
> 
> loadzone= in
> defaultzone = in
> 
> span=1,1,7,ccs,hdb3
> bchan=1-15
> dchan=16 
> bchan=17-31
> echocanceller=oslec,1-15,17-31
> 
> Is there anything missing or i am going wrong.. 
> 
> Help me out.
> 
> Thanks in advance...
> 
> 
> 
> -- 
> Regards,
> 
> Chandrakant Solanki
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


hehe ;) You are already built dahdi with oslec. You will not load
manually this module. 

Try!
Build dahdi, modprobe  (my module is wcfxo)

modprobe wcfxo:
(dmesg)
wcfxo :00:09.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17
wcfxo: DAA mode is 'FCC'



cat /etc/dahdi/system.conf

fxsks = 1
echocanceller =oslec,1-240
loadzone = ru
defaultzone = ru



dahdi_cfg -vv
DAHDI Tools Version - 2.2.0
*


Channel map:


Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01)


1 channels to configure.


Setting echocan for channel 1 to oslec




Hope it helps.. 

-- 
Best regards, Vince Mallow
xmpp: w...@jabber.slan.ru 
web: http://gentoo-way.blogspot.com


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
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Re: [asterisk-users] dahdi and oslec

[wins mallow]

On Wed, 2010-03-03 at 11:31 +0530, Chandrakant Solanki wrote:
> Hi All,
> 
> I have followed below steps to enable echo cancellation.
> 
> # cd /usr/src
> # wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2
> # tar xjf linux-2.6.28.tar.bz2
> # tar zxvf dahdi-linux-2.1.0.4.tar.gz
> # ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi
> # mkdir /usr/src/dahdi/drivers/staging
> # cp
>
-fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging
> # sed -i "s|#obj-m += dahdi_echocan_oslec.o|obj-m +=
> dahdi_echocan_oslec.o|" /usr/src/dahdi/drivers/dahdi/Kbuild
> # sed -i "s|#obj-m += ../staging/echo/|obj-m
> += ../staging/echo/|" /usr/src/dahdi/drivers/dahdi/Kbuild
> # echo 'obj-m += echo.o' > /usr/src/dahdi/drivers/staging/echo/Kbuild
> # cd /usr/src/dahdi
> # make
> # make install
> # cd /usr/src
> # tar zxvf dahdi-tools-2.1.0.2.tar.gz
> # cd /usr/src/dahdi-tools-2.1.0.2
> # ./configure
> # make
> # make install
> 
> # wget http://www.rowetel.com/ucasterisk/downloads/oslec-0.2.tar.gz
> # tar xvzf oslec-0.2.tar.gz
> # cd oslec-0.2
> # make
> # insmod kernel/oslec.ko
> 
> when i restart /etc/init.d/dahdi service it gives me following error
> in /var/log/message
> 
> Mar  3 11:06:37 server1 kernel: echo: exports duplicate symbol
> oslec_hpf_tx (owned by oslec)
> Mar  3 11:06:37 server1 modprobe: WARNING: Error inserting echo
> (/lib/modules/2.6.18-92.1.22.el5/staging/echo/echo.ko): Invalid module
> format 
> Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
> oslec_create
> Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
> oslec_update
> Mar  3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol
> oslec_free
> Mar  3 11:06:37 server1 modprobe: FATAL: Error inserting
> dahdi_echocan_oslec
> (/lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_echocan_oslec.ko):
> Unknown symbol in module, or unknown parameter (see dmesg) 
> 
> # cat /etc/dahdi/system.conf 
> 
> loadzone= in
> defaultzone = in
> 
> span=1,1,7,ccs,hdb3
> bchan=1-15
> dchan=16 
> bchan=17-31
> echocanceller=oslec,1-15,17-31
> 
> Is there anything missing or i am going wrong.. 
> 
> Help me out.
> 
> Thanks in advance...
> 
> 
> 
> -- 
> Regards,
> 
> Chandrakant Solanki
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


hehe ;) You are already built dahdi with oslec. You will not load
manually this module. 

Try!
Build dahdi, modprobe  (my module is wcfxo)

modprobe wcfxo:
(dmesg)
wcfxo :00:09.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17
wcfxo: DAA mode is 'FCC'



cat /etc/dahdi/system.conf

fxsks = 1
echocanceller =oslec,1-240
loadzone = ru
defaultzone = ru



dahdi_cfg -vv
DAHDI Tools Version - 2.2.0
*


Channel map:


Channel 01: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01)


1 channels to configure.


Setting echocan for channel 1 to oslec




Hope it helps.. 

-- 
Best regards, Vince Mallow
xmpp: w...@jabber.slan.ru 
web: http://gentoo-way.blogspot.com


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users