[asterisk-users] Digium Echo cancellation.

[DHAVAL INDRODIYA]

hi all,

any one know, about echo cancellation with digium card,

is it actually needed or it okay if we dont purchase because it increase
price which half of new card,

regards
Dhaval
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Re: [asterisk-users] Fw: app_swift issue

[ABBAS SHAKEEL]

Thanks Todd and Covici


Todd what Covici said  i have done it already but i was thinking it may be
slow

I also come to know from various resources that app_swift is good in
performance etc etc

however Todd  FYI you can use this link to accomplish what Covici said using
this link ===
http://samyantoun.50webs.com/asterisk/freepbx/clarkconnect/4.0/swift.htm


Any one else if know the issue what we are doing wrong or app_swift is doing
wrong to us ;)


-- 
Best Regards
Shakeel Abbas
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Re: [asterisk-users] netfilter conntrack mangling canreinvite?

[John A. Sullivan III]

On Tue, 2009-08-25 at 21:07 -0400, John A. Sullivan III wrote:
> Hello, all.  Since implementing an iptables firewall between the
> Asterisk PBX and several SIP phones, the Asterisk PBX ability to
> "reinvite" has been broken even when the phones are on the same network
> (i.e., no firewall between the phones).  We've been beating our heads
> against the wall thinking it was the complex rule set but it appears the
> issue is ip_conntrack_sip.
> 
> Before I drop another day into verifying this, may I ask if anyone else
> has had a similar problem and found a solution? It appears conntrack is
> rewriting the SDP so that the address is reverted to the PBX address.
> 
> Here are the relevant SDP portion of a reinvite captured on the PBX
> using tcpdump and displayed in Wireshark.  The PBX is at 172.x.x.8 and
> the phone is at 10.x.x.193:
> 
> Owner/Creator, Session Id (o): root 1417450700 1417450701 IN IP4
> 10.x.x.183
> Owner Address: 10.x.x.183
> Connection Information (c): IN IP4 10.x.x.183
> Connection Address: 10.x.x.183
> 
> Here is a similar sequence but captured from the phone itself:
> Owner/Creator, Session Id (o): root 595629021 595629022 IN IP4 172.x.x.8
> Owner Address: 172.x.x.8
> Connection Information (c): IN IP4 172.x.x.8
> Connection Address: 172.x.x.8
> 
> It would appear conntrack is incorrectly "fixed" the packet.
> 
> I noticed newer kernels have sip_direct_media and sip_direct_signalling
> options.  I don't know if those apply but they do not seem to be present
> in our CentOS 5.3 kernel.
> 
> I'll probably spend most of tomorrow confirming this hypothesis and
> investigating solutions so I'd be deeply appreciative for any
> time-saving advice.  Thanks - John
> 
The ip_nat_sip conntrack module was indeed the culprit.  Apparently this
can be fixed in newer kernels by setting the sip_direct_media=0 option
for ip_conntrack_sip in modprobe.conf.  However, since our CentOS 5.3
version of the kernel does not support this, we disabled ip_nat_sip and
returned responsibility for managing NAT to sip.conf.  Hope this helps
someone else - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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[asterisk-users] 2 Asterisk boxes : 1 can see the other, not vica versa

[jonas kellens]

Asterisk-server 1 :

register => hostedasterisk:pa...@domain2.biz

[BOX-YOCAN]
type=user ; incoming from YOCAN...
auth=rsa
inkeys=Key  ; pub key of user
secret=XXX  ; pass2 of YOCAN
context=from-BOX-YOCAN  ; incoming from YOCAN
trunk=yes
transfer=no
qualify=yes

[BOX-YOCAN]
type=peer ; outgoing to YOCAN...
host=dynamic  ; registers
auth=rsa
outkey=kEy  ; private secret key
username=hostedasterisk  ; username @ YOCAN
trunk=yes
transfer=no
qualify=yes

Asterisk-server 2 :

register => BOX-YOCAN:pa...@ip_hostedasterisk

[hostedasterisk]
type=user  ; incoming from Hosted Asterisk
auth=rsa
inkeys=HoStAsTkEy  ; pub key of AsteriskHosted
secret=XXX  ; pass1 of HostedAsterisk
context=from-HostAst  ; incoming from HostedAsterisk
trunk=yes
qualify=yes

[voipcenter]
type=peer
host=IP of HostedAsterisk
auth=rsa
outkey=Key  ; private secret key
username=BOX-YOCAN  ; username @ HostedAsterisk
trunk=yes
qualify=yes

Asterisk-server 2 (YOCAN) registers well to Asterisk-server 1
(AsteriskHosted) but registration of Asterisk-server 1 (AsteriskHosted)
is rejected by Asterisk-server 2 (YOCAN). Why is that ??

Jonas.
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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Loic Didelot]

It seems that the D-Channel goes down once the latency reaches 14ms.

Once I see this message the d-channel never comes back. Shouldnt zaptel
try to decrease the latency at some point?

[15742.188213] wcte12xp0: Missed interrupt. Increasing latency to 14 ms
in order to compensate.


Best regards,
Loïc.

On Wed, 2009-08-26 at 15:19 +0300, Tzafrir Cohen wrote:
> On Wed, Aug 26, 2009 at 02:13:13PM +0200, Loic Didelot wrote:
> 
> > Does zaptel really need to increase the latency and to change the
> > master?
> 
> Maybe it does. Maybe it doesn't. How about debugging the actual problem
> at hand (alarm was not properly cancelled, according to Asterisk)?
> 


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[asterisk-users] Company in Los Angeles looking for Asterisk & Network Administration/Maintenance Engineer

[James Lamanna]

Hi,
I work for a small VoIP/Internet service provider here in Southern
California and we are currently
looking for a Network Administrator who also knows Asterisk to add to
our support staff.
Some of your duties would include,
- Maintenance of current Linux servers
- Maintenance of current Asterisk servers
- Troubleshooting of Asterisk/server related issues
- Call routing provisioning

We are a very small company, so you will be given a significant amount of
latitude when it comes to administration tasks, and also be given the
opportunity to
grow with the company.
Being that we are very small company, you should be a self-starter
and also be able to contribute to enhancing the network as well,
through implementation of more proactive monitoring techniques and
whatever else you
may see fit.

Ideally this candidate will have had at least 5 years experience of Linux
server administration along with at least 1 year of Asterisk
administration and troubleshooting.

Cisco router/switch configuration/administration knowledge is a HUGE plus.

We are located in Pasadena, California, so ideally you would reside in the
Southern California area, but telecommuting options can be discussed.

Please email me directly for more details or any questions about the position.

Thanks.

James Lamanna
Warp2Biz, Inc.

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Re: [asterisk-users] TE4XXP: Version Synchronization Error!

[Kevin P. Fleming]

Joao Gomes Pereira wrote:

> Aug 26 17:58:16 catumbela kernel: VPM450: Not Present
> Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error!
> Aug 26 17:58:16 catumbela kernel: Completed startup!
> Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error!
> 
> 
> 
> 
> What could be worng with my dahdi  configuration?

This is not a configuration error; please contact Digium's Support
department.

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

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Re: [asterisk-users] LDAP Get for Asterisk 1.6.x

[Gavin Henry]

2009/8/24 David Klaverstyn :
> I’d appreciate it if someone could give me an answer to using LDAP in
> Asterisk 1.6.x

You can use res_config_ldap for storing Asterisk data in a directory
server for the realtime framework.

Thanks.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com

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Re: [asterisk-users] install the digium card TC400P howto?

[F6HQZ]

Hi Francois,

Here is Francois too.   :-)

Why to not ask to your Digium card provider ?

Example for Zaptel (fortunately same for Dahdi) :
via Linux console and dmesg :

Registered codec translator 'DTE Encoder' with 92 transcoders
(srcs=000c, dsts=0101)
Registered codec translator 'DTE Decoder' with 92 transcoders
(srcs=0101, dsts=000c)
Zaptel DTE (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12)


via CLI :

*CLI> transcoder show
0/0 encoders/decoders of 92 channels are in use.


change by this way via Linux console :

rmmmod wctc4xxp
modprobe wctc4xxp mode=g729

The module will be loaded with the "G729 exclusive mode" and 120 channels
will be available.

"dmesg" extract after that :

Registered codec translator 'DTE Encoder' with 120 transcoders
(srcs=000c, dsts=0100)
Registered codec translator 'DTE Decoder' with 120 transcoders
(srcs=0100, dsts=000c)
Zaptel DTE (G.729a) Transcoder support LOADED (firm ver = 6.12)
Found and successfully installed a Wildcard TC: Wildcard TC400P+TC400M


*CLI> transcoder show
0/0 encoders/decoders of 120 channels are in use.

Checks with 120 concurrent calls :

transcoder show
120/0 encoders/decoders of 120 channels are in use.


*CLI> core show channels
...SNIP...
240 active channels
120 active calls


*CLI> core show translation
 Translation times between formats (in milliseconds) for one second
of data
  Source Format (Rows) Destination Format (Columns)

  g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726
g722
 g723-   ---- -- -- --- 
   -
  gsm-   -222 21 4320   15
2-
 ulaw-   3-12 21 4120   15
2-
 alaw-   31-2 21 4120   15
2-
 g726aal2-   322- 21 4320   15
1-
adpcm-   3222 -1 4320   15
2-
 slin-   2111 1- 3219   14
1-
lpc10-   4333 32 -421   16
3-
 g729-   4113 32 5-21   16
3-
speex-   4333 32 54 -   16
3-
 ilbc-   5444 43 6522-
4-
 g726-   3221 21 4320
5--
 g722-   ---- -- -- --- 
   -


Best Regards,
Francois


  -Message d'origine-
  De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]de la part de BERGANZ
François
  Envoyé : mardi 25 août 2009 10:12
  À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Objet : Re: [asterisk-users] install the digium card TC400P howto?


  Ok.

  Now, I follow the digium documents.

  I have the TC400B-user-manual-1.pdf



  Where have I to insert the mode=g729 ?







  Chapter 3

  Configuration



  At this time no zaptel.conf or zapata.conf changes are necessary to
utilize

  this card. The ‘mode’ module parameter may be used to specify which

  complex codes are allowed.

  „ mode = mixed: This default option will enable 92 calls of G.729a or

  G.723.1 (5.3Kbit)

  „ mode = g729: This option will enable 96 calls of G.729a

  „ mode = g723: This default option will enable 92 calls of G.723.1

  (5.3Kbit)







  Cordialement,

  BERGANZ François





  http://www.acropolistelecom.net

  P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.



  De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Steve Totaro
  Envoyé : lundi 24 août 2009 19:17
  À : Asterisk Users Mailing List - Non-Commercial Discussion
  Objet : Re: [asterisk-users] install the digium card TC400P howto?





  On Mon, Aug 24, 2009 at 1:06 PM, Sean Bright 
wrote:

  Steve Totaro wrote:
  > "No hardware timing source found in /proc/dahdi, loading dahdi_dummy"
  > would make me think it is not loading correctly.

  The TC400P is a transcoder card.  It is not a timing source.

  --

  Sean Bright
  sean.bri...@gmail.com


  Silly me.  I forgot about those overpriced transcoder cards.

  Dollar for dollar, I will go for bogomips!


  --
  Thanks,
  Steve Totaro
  +18887771888 (Toll Free)
  +12409381212 (Cell)
  +12024369784 (Skype)

Analyse effectuée par AVG - www.avg.fr
Version: 8.5.375 / Base de données virale: 270.13.67/2326 - Date: 08/25/09
18:07:00
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Re: [asterisk-users] Fw: app_swift issue

[covici]

You could just use the dialplan and a small agi or something to create
the wav file and then play the file -- at least this is what I did.

Todd Fulton  wrote:

> Hi Shakeel,
> 
> I had the same problem building app_swift (1.6..) myself and searched the web 
> far-and-wide for a solution.  I eventually contacted Darren Sessions -- who 
> was maintaining that plug-in -- about a month ago.  He was involved in 
> another project and said he might be able get to it after a few weeks.  But, 
> since then, his website http://www.darrensessions.com/ has gone out of 
> comission.
> 
> I think we may be on our own on this one!
> 
> 
> Todd
> 
> 
> 
> - Forwarded Message 
> From: Todd Fulton 
> To: Todd Fulton 
> Sent: Wednesday, August 26, 2009 10:59:34 AM
> Subject: [asterisk-users] app_swift issue
> 
> 
> Hello 
> 
> I have installed cepstral  It works woderfull using an agi script but 
> . 
> when i try to use Swift("say this") is Dial plan  I get the error 
> 
> [Aug 26 12:30:18] WARNING[7420]: pbx.c:3167 pbx_extension_helper: No 
> application 'Swift' for extension (actdemo, 123, 2) 
> 
> 
> 
> Now i come to know to install app_swift 
> 
> 
> Here is the issue... 
> 
> when i try to execute make command on app_swift-1.6.2 
> 
> I get the following error 
> 
> [r...@asterisk app_swift-1.6.2]# make 
> gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC   -c -o 
> app_swift.o app_swift.c 
> app_swift.c: In function ‘engine’: 
> app_swift.c:402: error: incompatible types in assignment 
> app_swift.c: In function ‘load_module’: 
> app_swift.c:546: error: ‘AST_MODULE’ undeclared (first use in this function) 
> app_swift.c:546: error: (Each undeclared identifier is reported only once 
> app_swift.c:546: error: for each function it appears in.) 
> make: *** [app_swift.o] Error 1 
> 
> 
> 
> Now i am thinking to edit app_swift.c but AST_MODULE is not defined in 
> app_swift.c 
> 
> i commented this line ""//#define AST_MODULE "app_swift"" 
> 
> but in vain  Please help 
> 
> static int load_module(void) 
> { 
> int res; 
> const char *t = NULL; 
> struct ast_config *cfg; 
> struct ast_flags config_flags = { 0 }; 
> 
> // Set defaults 
> cfg_buffer_size = 65535; 
> cfg_goto_exten = 0; 
> strncpy(cfg_voice, "David-8kHz", sizeof(cfg_voice)); 
> 
> res = ast_register_application(app, engine, synopsis, descrip); 
> cfg = ast_config_load(SWIFT_CONFIG_FILE, config_flags); 
> 
> if (cfg) { 
> if ((t = ast_variable_retrieve(cfg, "general", "buffer_size"))) { 
> cfg_buffer_size = atoi(t); 
> ast_log(LOG_DEBUG, "Config buffer_size is %d\n", 
> cfg_buffer_size); 
> } 
> if ((t = ast_variable_retrieve(cfg, "general", "goto_exten"))) { 
> if (!strcmp(t, "yes")) 
> cfg_goto_exten = 1; 
> else 
> cfg_goto_exten = 0; 
> ast_log(LOG_DEBUG, "Config goto_exten is %d\n", cfg_goto_exten); 
> } 
> 
> ast_config_destroy(cfg); 
> 
> } else { 
> ast_log(LOG_NOTICE, "Failed to load config\n"); 
> } 
> 
> return res; 
> } 
> 
> char *description(void) 
> { 
> return tdesc; 
> } 
> 
> 
> 
> #define AST_MODULE "app_swift" 
> 
> AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Cepstral Swift TTS 
> Application"); 
> 
> 
> 
> 
> 
> 
> 
> 
> -- 
> Best Regards
> Shakeel Abbas
> 
> Alternatives:
> 
> 
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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[asterisk-users] Youmail DropBox - Email interface to Youmail transcription service

[Karl Fife]

If you haven't heard, Youmail has recently created a drop-box email interface 
for their transcription service.  As of right now, it's free, but whether free 
or paid, this may be good news for those on this list who want to annex 
transcripion functionality to your current mix of voicemail offerings.

For example:
We (like many) use Asterisk to integrate the voicemail of mobile, legacy PBX, 
and 'normal' Asterisk mailboxes to create a unified messaging interface.  
DropBox will allow us to tack on a transcription function with trivial effort.  
Before the DropBox interface you'd have needed to kludge up something like a 
drill-down bot to play the message to the youmail service in baseband audio.  

Imagine the efficiency of doing something like this: 
Re-relay [sic] the youmail transcription with the addtion of a special 
single-use URL (or even special Phone number), which, when visited by the 
subscriber (or momentarily called with the CallerID of the subscriber) would 
trigger a change to the status of that message back at the Asterisk mothership. 
 In this way, the end user (if satisfied with the transcription) could opt out 
of the need to manually visit the audio interface of their voicmail box for a 
properly transcribed message.  

We've been impressed with the performance of youmail within the silo of it's 
normal customer-facing UI, but we have not yet had a chance to test drive 
DropBox as integrated with Asterisk.  We plan to do so shortly after we put out 
a few fires and free up some creative bandwidth. 

Youmail on geting started with DropBox (FAQ):
http://www.youmail.com/help/view/faq_dropoff_start

-Karl 


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[asterisk-users] 408 error

[David Shauger]

An associate is having the following issue with his Elastix server:

I am trying to get a remote extension to register via a site-site VPN  
connection. The SIP phone/client comes back with a 408 error. After  
much troubleshooting, I decided to install a fresh new copy of Elastix  
with FreePBX all updated to current releases. I configured this to  
have the same IP address as the "broken" PBX server (having shut down  
the old one). I added a new extension in this system and the remote  
phone registers fine. I did this to make sure I was not having other  
issues related to the firewall/router/VPN. So I assume this removes  
that from play.


I have looked though many of the /etc/asterisk config files to see  
what the differences are of the working and not working servers. I  
cannot find anything that stands out. I tried backing up the config  
files from the old server and restoring them on the new server to see  
if it might be a linux config issue. However the restored config from  
the old server onto the new server failed also.


Any ideas...please.


David Shauger
Vice President

Sollos Technology Solutions

678-317-9444 - voice
404-886-7603 - cell
772-679-5830 - fax
d...@sollos.com
http://www.sollos.com/

This email has been certified by Thawte
Email certification helps prevent identity theft
Virus scanning provided by Clam Antivirus





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Re: [asterisk-users] Multiple user registration ...

[Mauro Sergio Ferreira Brasil]

Thanks again Barry for the help and attention.
Thanks for wishing me lucky as well... If we insist on this road I'll 
need it for sure :-).

I can't agree more with your position, and I'll try to be sure our 
commercial demands can't be acchieved with normal approaches before 
adventuring on such path.

Best regards,
Mauro.



Barry L. Kline escreveu:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
>   
> Well, our phones at home are probably analog and can be connected in
> parallel.  Unfortunately, VoIP phones are a different matter and need to
> be identified individually.
>
> I guess I don't get the problem your commercial side is having with this
> concept.  You can produce the same result doing things within the
> constraints of SIP using the features built into Asterisk.
>
> Doing what you want may be possible with a bunch of contortions, but
> it's going to be an unnatural act fraught with tons of unexpected
> behavior.  If you do get it working the way you describe you'll likely
> be doing so because of a side-effect behavior in a GIVEN version of
> Asterisk.  The moment you change versions, the side effect may or may
> not be the same and you may find yourself in the same trouble.
>
> I can't offer anything more to help you except to wish you the best of
> luck.  You're going to need it.
>
> Barry
>
>
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.5 (GNU/Linux)
>
> iD8DBQFKlWonCFu3bIiwtTARAu1WAJ0eS2Eh6n6Tici9eDA82UIesuozNACaA9yi
> jT8u2aZfUHcSXGvJnc1FDEI=
> =VQhJ
> -END PGP SIGNATURE-
>
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>
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   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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Re: [asterisk-users] Multiple user registration ...

[Mauro Sergio Ferreira Brasil]

Thanks again Elliot for everything.
Considering our needs to develop a proprietary ARA driver, I think it's 
possible to use it and "make Asterisk believe" that additional 
registrations of same SIP device are in fact different device registrations.

BUT, and yes it's a big "BUT", we will end with an Asterisk version a 
litle hacked and even if we get this working on some version now, it 
doesn't give us any guarantee that it will in future.
Anyway, I've put this question here just to be sure no one has already 
made such a thing before, and how odd is it.

I'll take care and don't use "-" on sip device names... thanks for that too.

Best regards,
Mauro.



Elliot Otchet escreveu:
> Your first example illustrates why having multiple devices registered as the 
> same entity is a bad idea.  It is impossible to differentiate between each 
> device when you have multiple registering as the same entity.
>
> My users also really like setting up rules per device/per caller.  When you 
> treat a group of devices as one, you make it really hard to do that.
>
> On your theoretical "virtual" devices in Asterisk - you either have a device 
> or you don't.  The device will need to register in order to receive a call, 
> so if you're expecting to do some magic on the registration to have a user 
> who registers with the credentials of user 101 and be assigned to user 
> 101-001, you'll be disappointed in the results.
>
> Also, you'll want to steer away from using hyphens in your sip device names.  
> Hyphens are used in the SIP channel driver for a special purpose and using 
> them in your device names may cause problems.  See 
> http://www.digium.com/handbook-draft.pdf page 19 for more info.  If you're 
> looking for a good separator, try using the underscore (_) character instead.
>
> All that being said, if you want to register multiple devices with a single 
> set of credentials, you might want to check out a SIP Proxy instead of 
> Asterisk's SIP B2BUA.  Some can handle multiple registrations with a single 
> set of credentials quite nicely.
>
> Regards,
>
> Elliot
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio 
> Ferreira Brasil
> Sent: Wednesday, August 26, 2009 12:19 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Multiple user registration ...
>
> Hi Elliot, and thanks for the reply.
>
> I'm not completely sure you've considered that the SIP users registered
> on all devices are the same.
> Have you ?
>
> I mean...
> How will I use "Dial" command with a sequence of same devices, like:
> Dial(SIP/101&SIP/101&SIP/101), for example ?
>
> That's why we are testing the possibility to create "virtual" devices on
> subsequent registrations, so we can at the end make something like:
> Dial(SIP/101&SIP/101-001&SIP/101-002) if someone dials to SIP/101.
> Note: SIP/101-001 and SIP/101-002 don't really exist. They will be
> provided by our ARA driver to allow the multiple device ringing.
>
> Thanks and best regards,
> Mauro.
>
>
>
> Elliot Otchet escreveu:
>   
>> Is your goal here to have multiple devices ring when an extension is dialed 
>> and the first one to answer take the call?
>>
>> If so, see the Dial command 
>> Dial(Technology/resource&Technology/resource&Technology/resource...[|timeout][|options][|URL]).
>>   When multiple technology/resource entries are listed, the first one to 
>> answer will take the call.  That accomplishes your goal, if I understand you 
>> correctly.
>>
>> The nice part about doing it this way (with each device independently 
>> registered) is that you gain a substantial amount of granularity in 
>> controlling where calls go and you don't have to find creative ways (read: 
>> unsupported) to trick Asterisk or endpoints.
>>
>> If you're developing your own GUI to have people set up their devices, you 
>> can easily create a wizard that walks them through setting up each device 
>> and associating them together through either channel variables or other 
>> tables in a database.
>>
>> I use this methodology in 1.4 and it works quite reliably.  For a good 
>> reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or 
>> from your Asterisk console try: 'core show application dial'
>>
>> It's not perfect because you can have devices that do funny things with a 
>> SIP INVITE, but in most cases it works very well.
>>
>> Regards,
>>
>> Elliot
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>>
>>
>> 
>
> --
> __At.,
>_
>
> *Technology and Quality on Information*
> Mauro Sérgio Ferreira Brasil
> Coorde

[asterisk-users] Fw: app_swift issue

[Todd Fulton]

Hi Shakeel,

I had the same problem building app_swift (1.6..) myself and searched the web 
far-and-wide for a solution.  I eventually contacted Darren Sessions -- who was 
maintaining that plug-in -- about a month ago.  He was involved in another 
project and said he might be able get to it after a few weeks.  But, since 
then, his website http://www.darrensessions.com/ has gone out of comission.

I think we may be on our own on this one!


Todd



- Forwarded Message 
From: Todd Fulton 
To: Todd Fulton 
Sent: Wednesday, August 26, 2009 10:59:34 AM
Subject: [asterisk-users] app_swift issue


Hello 

I have installed cepstral  It works woderfull using an agi script but 
. 
when i try to use Swift("say this") is Dial plan  I get the error 

[Aug 26 12:30:18] WARNING[7420]: pbx.c:3167 pbx_extension_helper: No 
application 'Swift' for extension (actdemo, 123, 2) 



Now i come to know to install app_swift 


Here is the issue... 

when i try to execute make command on app_swift-1.6.2 

I get the following error 

[r...@asterisk app_swift-1.6.2]# make 
gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC   -c -o 
app_swift.o app_swift.c 
app_swift.c: In function ‘engine’: 
app_swift.c:402: error: incompatible types in assignment 
app_swift.c: In function ‘load_module’: 
app_swift.c:546: error: ‘AST_MODULE’ undeclared (first use in this function) 
app_swift.c:546: error: (Each undeclared identifier is reported only once 
app_swift.c:546: error: for each function it appears in.) 
make: *** [app_swift.o] Error 1 



Now i am thinking to edit app_swift.c but AST_MODULE is not defined in 
app_swift.c 

i commented this line ""//#define AST_MODULE "app_swift"" 

but in vain  Please help 

static int load_module(void) 
{ 
int res; 
const char *t = NULL; 
struct ast_config *cfg; 
struct ast_flags config_flags = { 0 }; 

// Set defaults 
cfg_buffer_size = 65535; 
cfg_goto_exten = 0; 
strncpy(cfg_voice, "David-8kHz", sizeof(cfg_voice)); 

res = ast_register_application(app, engine, synopsis, descrip); 
cfg = ast_config_load(SWIFT_CONFIG_FILE, config_flags); 

if (cfg) { 
if ((t = ast_variable_retrieve(cfg, "general", "buffer_size"))) { 
cfg_buffer_size = atoi(t); 
ast_log(LOG_DEBUG, "Config buffer_size is %d\n", 
cfg_buffer_size); 
} 
if ((t = ast_variable_retrieve(cfg, "general", "goto_exten"))) { 
if (!strcmp(t, "yes")) 
cfg_goto_exten = 1; 
else 
cfg_goto_exten = 0; 
ast_log(LOG_DEBUG, "Config goto_exten is %d\n", cfg_goto_exten); 
} 

ast_config_destroy(cfg); 

} else { 
ast_log(LOG_NOTICE, "Failed to load config\n"); 
} 

return res; 
} 

char *description(void) 
{ 
return tdesc; 
} 



#define AST_MODULE "app_swift" 

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Cepstral Swift TTS 
Application"); 








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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Loic Didelot]

I disabled one network interface which shared the IRQ with the wildcard.
Lets see if it misses still so many IRQs.

Best regards,
Loïc.


On Wed, 2009-08-26 at 20:56 +0300, Tzafrir Cohen wrote:
> On Wed, Aug 26, 2009 at 05:07:58PM +0200, Loic Didelot wrote:
> > Yes, I see the clearing, but the InAlarm flag stays to 1. 
> 
> So where exactly do you see it clearing? What line in the log?
> 
> > Is there a way
> > to restart zaptel without restarting the server.
> > 
> > I tried restarting asterisk which did not help.
> 
> That's odd.
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Tzafrir Cohen]

On Wed, Aug 26, 2009 at 05:07:58PM +0200, Loic Didelot wrote:
> Yes, I see the clearing, but the InAlarm flag stays to 1. 

So where exactly do you see it clearing? What line in the log?

> Is there a way
> to restart zaptel without restarting the server.
> 
> I tried restarting asterisk which did not help.

That's odd.

-- 
   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] ACD, call barge, recording

[Steve Edwards]

On Wed, 26 Aug 2009, Tilghman Lesher wrote:

> Voip-info is often wrong, depends upon 3rd party modules not distributed 
> with Asterisk, and leads to more confusion, rather than clarifying 
> issues.

All true, yet sometimes it is still the best resource.

-- 
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] TE4XXP: Version Synchronization Error!

[Joao Gomes Pereira]

Im also getting these errors:




Aug 26 17:58:17 catumbela kernel: dahdi: Registered tone zone 0 (United 
States / North America)
Aug 26 17:58:17 catumbela kernel: About to enter startup!
Aug 26 17:58:17 catumbela kernel: TE4XXP: Version Synchronization Error!
Aug 26 17:58:17 catumbela last message repeated 127 times
Aug 26 17:58:17 catumbela kernel: TE4XXP: Span 1 configured for CAS/HDB3
Aug 26 17:58:17 catumbela kernel: TE4XXP: Version Synchronization Error!
Aug 26 17:58:17 catumbela last message repeated 11 times
Aug 26 17:58:17 catumbela dahdi: Running dahdi_cfg:  succeeded
Aug 26 17:58:18 catumbela kernel: TE4XXP: Version Synchronization Error!
Aug 26 17:58:18 catumbela last message repeated 21 times
Aug 26 17:58:18 catumbela kernel: wct4xxp: Setting yellow alarm on span 1
Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error!
Aug 26 17:58:16 catumbela last message repeated 67 times
Aug 26 17:58:16 catumbela kernel: SPAN 1: Primary Sync Source
Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error!
Aug 26 17:58:16 catumbela last message repeated 2 times
Aug 26 17:58:16 catumbela kernel: VPM400: Not Present
Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error!
Aug 26 17:58:16 catumbela last message repeated 37 times
Aug 26 17:58:16 catumbela kernel: VPM450: Not Present
Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error!
Aug 26 17:58:16 catumbela kernel: Completed startup!
Aug 26 17:58:16 catumbela kernel: TE4XXP: Version Synchronization Error!




What could be worng with my dahdi  configuration?
Thanks
regards
Joao Pereira




Joao Gomes Pereira escreveu:
> Hello to all
> I'm using asterisk 1.4 and dahdi.
> I had everything working fine, and I could place calls through my R2 
> channel.
> But now the channel is always "RED" and Im getting this error message:
> 
> TE4XXP: Version Synchronization Error!
> 
> 
> Here is my chan_dahdi.conf--
> 
> 
> [channels]
> language=en
> context=incomingr2
> 
> signalling=mfcr2
> 
> mfcr2_variant=ar
> 
> 
> switchtype=national
> 
> mfcr2_get_ani_first=no
> mfcr2_max_ani=20
> mfcr2_max_dnis=9
> mfcr2_category=national_subscriber
> 
> channel =>1-15,17-30
> 
> here is my dahdi config: 
> 
> span=1,1,0,cas,hdb3
> cas=1-15:1101
> cas=17-30:1101
> dchan=16
> 
> ---
> 
> What could be the problem?
> Why was this working fine and now the channel is RED?
> 
> Thanks
> Regards
> Joao Pereira
> 
> 

-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt

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Re: [asterisk-users] Multiple user registration ...

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mauro Sergio Ferreira Brasil wrote:


> I totally agree with you that this is an unnatural behavior, but I have 
> to agree as well with our commercial staff because their vision was 
> naturaly translated from our telephony world (we don't have a different 
> ID - telephone number - to each phone we have home, right ?).

Well, our phones at home are probably analog and can be connected in
parallel.  Unfortunately, VoIP phones are a different matter and need to
be identified individually.

I guess I don't get the problem your commercial side is having with this
concept.  You can produce the same result doing things within the
constraints of SIP using the features built into Asterisk.

Doing what you want may be possible with a bunch of contortions, but
it's going to be an unnatural act fraught with tons of unexpected
behavior.  If you do get it working the way you describe you'll likely
be doing so because of a side-effect behavior in a GIVEN version of
Asterisk.  The moment you change versions, the side effect may or may
not be the same and you may find yourself in the same trouble.

I can't offer anything more to help you except to wish you the best of
luck.  You're going to need it.

Barry



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Re: [asterisk-users] Multiple user registration ...

[Elliot Otchet]

Your first example illustrates why having multiple devices registered as the 
same entity is a bad idea.  It is impossible to differentiate between each 
device when you have multiple registering as the same entity.

My users also really like setting up rules per device/per caller.  When you 
treat a group of devices as one, you make it really hard to do that.

On your theoretical "virtual" devices in Asterisk - you either have a device or 
you don't.  The device will need to register in order to receive a call, so if 
you're expecting to do some magic on the registration to have a user who 
registers with the credentials of user 101 and be assigned to user 101-001, 
you'll be disappointed in the results.

Also, you'll want to steer away from using hyphens in your sip device names.  
Hyphens are used in the SIP channel driver for a special purpose and using them 
in your device names may cause problems.  See 
http://www.digium.com/handbook-draft.pdf page 19 for more info.  If you're 
looking for a good separator, try using the underscore (_) character instead.

All that being said, if you want to register multiple devices with a single set 
of credentials, you might want to check out a SIP Proxy instead of Asterisk's 
SIP B2BUA.  Some can handle multiple registrations with a single set of 
credentials quite nicely.

Regards,

Elliot


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio 
Ferreira Brasil
Sent: Wednesday, August 26, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple user registration ...

Hi Elliot, and thanks for the reply.

I'm not completely sure you've considered that the SIP users registered
on all devices are the same.
Have you ?

I mean...
How will I use "Dial" command with a sequence of same devices, like:
Dial(SIP/101&SIP/101&SIP/101), for example ?

That's why we are testing the possibility to create "virtual" devices on
subsequent registrations, so we can at the end make something like:
Dial(SIP/101&SIP/101-001&SIP/101-002) if someone dials to SIP/101.
Note: SIP/101-001 and SIP/101-002 don't really exist. They will be
provided by our ARA driver to allow the multiple device ringing.

Thanks and best regards,
Mauro.



Elliot Otchet escreveu:
> Is your goal here to have multiple devices ring when an extension is dialed 
> and the first one to answer take the call?
>
> If so, see the Dial command 
> Dial(Technology/resource&Technology/resource&Technology/resource...[|timeout][|options][|URL]).
>   When multiple technology/resource entries are listed, the first one to 
> answer will take the call.  That accomplishes your goal, if I understand you 
> correctly.
>
> The nice part about doing it this way (with each device independently 
> registered) is that you gain a substantial amount of granularity in 
> controlling where calls go and you don't have to find creative ways (read: 
> unsupported) to trick Asterisk or endpoints.
>
> If you're developing your own GUI to have people set up their devices, you 
> can easily create a wizard that walks them through setting up each device and 
> associating them together through either channel variables or other tables in 
> a database.
>
> I use this methodology in 1.4 and it works quite reliably.  For a good 
> reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or 
> from your Asterisk console try: 'core show application dial'
>
> It's not perfect because you can have devices that do funny things with a SIP 
> INVITE, but in most cases it works very well.
>
> Regards,
>
> Elliot
>
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>
>

--
__At.,
   _

*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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This message is intended only for the use of the individual (s) or entity to 
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the reader of this message is not the intended recipient, you are hereby 
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Re: [asterisk-users] Multiple user registration ...

[Mauro Sergio Ferreira Brasil]

Hi Barry, and thanks for the reply!

This was the first question I've made on meeting yesterday to decide 
about this facility.
Having me here today making this question should give you an idea of the 
level of acceptance of my suggestion :-).

Anyway, the idea is really try to make it work with only one SIP user.

I totally agree with you that this is an unnatural behavior, but I have 
to agree as well with our commercial staff because their vision was 
naturaly translated from our telephony world (we don't have a different 
ID - telephone number - to each phone we have home, right ?).

So, I thank you for your handy "Dial" approach, which will be easier 
than the queue approach I was considering before.
Given that I'll acchieve the "virtual devices" running.

Considering my annoying insistence on work with just one SIP user, do 
you have any helpfull thoughts to share that can help me out ?

Best regards,
Mauro.


Barry L. Kline escreveu:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
>   
> Instead of trying to make Asterisk do this unnatural act, why not
> register each device with a separate id, then use the dial function to
> call all of them?
>
> e.g.exten => 122,1,Dial(SIP/1&SIP/2&SIP/3)
>
> You could use some creating scripting to decide which devices to ring
> based on the dialed extension.
>
> Barry
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.5 (GNU/Linux)
>
> iD8DBQFKlU65CFu3bIiwtTARAu0DAJ4szfX1dp/BNZojIKhgIL/tIhkjvQCeLXCf
> A+Dys6+LgrNhL/zQpU8Vuwk=
> =1Y6q
> -END PGP SIGNATURE-
>
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>
>   

-- 
__At.,  
   
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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Re: [asterisk-users] User-invoked call restrictions

[Steve Edwards]

Un-top-posting...


On Wed, Aug 26, 2009 at 4:39 AM, David A. Bandel


Had a request from a customer: ?is it possible for a customer, using a 
password to restrict others from making long distance/cell calls? That 
is, the user set a level of service?


[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roberto 
Piola Sent: Wednesday, August 26, 2009 6:37 AM


Hint: an AGI application that looks into a database for passwords, and 
the decides, according to the prefix, if the call is allowed or not


On Wed, 26 Aug 2009, Danny Nicholas wrote:


You could also do this in the dialplan without AGI.


If this is an "only the boss can make $$$ calls" and the password is 
unlikely to change, hardcode it in the dialplan.


Next step up is to store the password in the Asterisk DB.

Next step up would be an external database (MySQL) and a web page for 
frequent updates. Personally, at this stage I would use an AGI to keep the 
dialplan from getting ugly.


--
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] Follow me IVR sounds

[Steve Edwards]

On Wed, 26 Aug 2009, James Mutuku wrote:

> thanks danny for the reply. I am looking into using flite to read out the
> prompts. if i ma ask...are there other voices other than the mechanical
> robotic male voice available for flite.? I have searched over the internet
> and I can't seem to find any

With my limited TTS experience, I find Cepstral's Allison's voice font to 
be more than acceptable. Sometimes somewhat mechanical, sometimes 
indistinguishable from her "live" prompts supplied with Asterisk.

If you can, create and normalize your prompts offline for better 
performance.

-- 
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] Multiple user registration ...

[Mauro Sergio Ferreira Brasil]

Hi Elliot, and thanks for the reply.

I'm not completely sure you've considered that the SIP users registered 
on all devices are the same.
Have you ?

I mean...
How will I use "Dial" command with a sequence of same devices, like: 
Dial(SIP/101&SIP/101&SIP/101), for example ?

That's why we are testing the possibility to create "virtual" devices on 
subsequent registrations, so we can at the end make something like: 
Dial(SIP/101&SIP/101-001&SIP/101-002) if someone dials to SIP/101.
Note: SIP/101-001 and SIP/101-002 don't really exist. They will be 
provided by our ARA driver to allow the multiple device ringing.

Thanks and best regards,
Mauro.



Elliot Otchet escreveu:
> Is your goal here to have multiple devices ring when an extension is dialed 
> and the first one to answer take the call?
>
> If so, see the Dial command 
> Dial(Technology/resource&Technology/resource&Technology/resource...[|timeout][|options][|URL]).
>   When multiple technology/resource entries are listed, the first one to 
> answer will take the call.  That accomplishes your goal, if I understand you 
> correctly.
>
> The nice part about doing it this way (with each device independently 
> registered) is that you gain a substantial amount of granularity in 
> controlling where calls go and you don't have to find creative ways (read: 
> unsupported) to trick Asterisk or endpoints.
>
> If you're developing your own GUI to have people set up their devices, you 
> can easily create a wizard that walks them through setting up each device and 
> associating them together through either channel variables or other tables in 
> a database.
>
> I use this methodology in 1.4 and it works quite reliably.  For a good 
> reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or 
> from your Asterisk console try: 'core show application dial'
>
> It's not perfect because you can have devices that do funny things with a SIP 
> INVITE, but in most cases it works very well.
>
> Regards,
>
> Elliot
>
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-- 
__At.,  
   
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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[asterisk-users] TE4XXP: Version Synchronization Error!

[Joao Gomes Pereira]

Hello to all
I'm using asterisk 1.4 and dahdi.
I had everything working fine, and I could place calls through my R2 
channel.
But now the channel is always "RED" and Im getting this error message:

TE4XXP: Version Synchronization Error!


Here is my chan_dahdi.conf--


[channels]
language=en
context=incomingr2

signalling=mfcr2

mfcr2_variant=ar


switchtype=national

mfcr2_get_ani_first=no
mfcr2_max_ani=20
mfcr2_max_dnis=9
mfcr2_category=national_subscriber

channel =>1-15,17-30

here is my dahdi config: 

span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-30:1101
dchan=16

---

What could be the problem?
Why was this working fine and now the channel is RED?

Thanks
Regards
Joao Pereira


-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt

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Re: [asterisk-users] Breaking news, but what happened? 11.000 channels on one server

[David Backeberg]

On Wed, Aug 26, 2009 at 11:35 AM, Peder wrote:
> On a side note, why do people buy high-end servers like IBM or HP and then
> put in crappy switches, like Dell or Netgear and then wonder why performance
> is bad?  That's like buying a BMW 7 series and then using the cheapest 87
> octane gas you can find.  If you want to build a good network, buy good
> network equipment like Cisco, Extreme, Foundry or some of the other high end
> manufacturers.  DLink, Linksys, Netgear and Dell are all low end consumer
> grade.  No matter how they may try and sell it, that's what it is.  It is
> fine for that, but it is not enterprise grade equipment.  I just ran across
> a customer that had a Cisco Catalyst 4000 with an uptime of 1500 days (4+
> years).  Try and get that with Linksys or Netgear.

I actually agree with your argument, and I've personally told people
before to chuck their Dell switch overboard and replace it with a
crossover cable.

However, I sure hope that four-years-ago IOS Cisco switch isn't on the
edge. That's way too long to not upgrade your IOS.

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Re: [asterisk-users] Breaking news, but what happened? 11.000 channels on one server

[Peder]

You are thinking IP (layer3), not mac address (layer2 - ethernet switching).
Bonding is general a poor choice of wording for multiple Ethernet
connections as an individual connection won't use both links.  The way most
NIC's and switches do bonding is that they hash the source and destination
mac address and odd packets go over one link and even packets go over the
other (assuming two links).  So if there are two machines talking, they will
flood one link and the other will be empty.  If there are 100 machines
talking to one machine, then it will be fairly even balancing.  If the PBX
is behind a router or firewall, then it will see all external IPs as one mac
(the router or firewall), so you again will get one link saturated and one
empty.  Your best bet is 10G if you can afford it.

On a side note, why do people buy high-end servers like IBM or HP and then
put in crappy switches, like Dell or Netgear and then wonder why performance
is bad?  That's like buying a BMW 7 series and then using the cheapest 87
octane gas you can find.  If you want to build a good network, buy good
network equipment like Cisco, Extreme, Foundry or some of the other high end
manufacturers.  DLink, Linksys, Netgear and Dell are all low end consumer
grade.  No matter how they may try and sell it, that's what it is.  It is
fine for that, but it is not enterprise grade equipment.  I just ran across
a customer that had a Cisco Catalyst 4000 with an uptime of 1500 days (4+
years).  Try and get that with Linksys or Netgear.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Tuesday, August 25, 2009 8:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Breaking news, but what happened? 11.000
channels on one server

On Tue, Aug 25, 2009 at 12:50 PM, John A. Sullivan
III wrote:
> You don't necessarily need a switch to support it.  One can use alb mode
> in Linux on any old switch and it works reasonably well other than for
> some excessive ARP traffic.  However, as we found out the hard way when
> building our Nexenta SAN, bonding works very well with many-to-many
> traffic but does very little to boost one-to-one network flows.  They
> will all collapse to the same pair of NICs in most scenarios and, in the
> one mode where they do not, packet sequencing issues will reduce the
> bandwidth to much less than the sum of the connections.  Take care -

Your claims make sense for a typical
Machine A has one IP address
Machine B has one IP address

And there is only one route between A and B. In this scenario, yes,
all calls take same route.

But what about giving each machine two addresses, two routes. And
halve your calls between the two paths between the same systems.
Doesn't this get around your problem, and allow you a chance to
saturate double the number of interfaces?

If you have four interfaces (as my new boxes do), lather, rinse,
repeat. Anybody have any reason why spreading the bandwidth across
multiple routes wouldn't get around this problem?

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Re: [asterisk-users] ACD, call barge, recording

[Tilghman Lesher]

On Wednesday 26 August 2009 00:30:57 Lee, John (Sydney) wrote:
> >  1) Can ACD (Automatic Call Distribution) service work with asterisk, and
> > how to set up ACD in asterisk ?
>
> You can (and it is better to) write your own code in Asterisk.

I'm not sure why.  The Queue app works fine.

> >  2) How call barging can set up in asterisk ?
>
> There is a zap barge cmd - not sure if this is what you want.

ChanSpy is probably what he wants.

> > 3) How call recording can set up in asterisk?
>
> You can set up one-touch recording pretty easily.

Note that that's in features.conf.  If the OP wants to simply
record all calls, using MixMonitor is probably a better solution.

> Please check voip-info.org

Voip-info is often wrong, depends upon 3rd party modules not distributed
with Asterisk, and leads to more confusion, rather than clarifying issues.

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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[asterisk-users] Swift application and DTMF

[srinivas Antarvedi]

Hello users,

i have successfully installed the cepstral voice and in the text only mode
its working fine

when i swift applicaiton in dtmf mode like

exten =>111,1,Swift(hello user| 5000|1)
exten =>111,n,NoOp(dtmf is ${SWIFT_DTMF})
exten => 111,n,Hangup()

case1:
when i am listening to the hello user prompt if i press any key
1,2,3,4,5,6,7,8,9,0,*,#
i am getting the ${SWIFT_DTMF } value as
1xx  <-- if i press 1
2xx  <-- if i press 2

and this is the same for all other digits including 0,#,* keys
and the prompt stops

i am getting following in my asterisk CLI


app_swift.c:453 engine:DTMF=#147987812  <--when i press #
app_swift.c:453 engine:DTMF=7147987812  <--when i press 7


case2:
when i pressed the above digits after the prompt finishes

i get   1 <-- if i press 1
  2 <--if i press 2
 etc.

 except  there is no DTMF detection
 for numbers 0(zero),#,*
and i am getting the following in my asterisk CLI

app_swift.c:453 engine:DTMF=7  <--when i press 7
app_swift.c:482 engine: No DTMF  <--when i press #,0,*

Please help me out

Thanks in advance
srinivas antarvedi
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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Loic Didelot]

Yes, I see the clearing, but the InAlarm flag stays to 1. Is there a way
to restart zaptel without restarting the server.

I tried restarting asterisk which did not help.

Loic


On Wed, 2009-08-26 at 13:27 +0300, Tzafrir Cohen wrote:
> On Wed, Aug 26, 2009 at 11:53:25AM +0200, Loic Didelot wrote:
> > Here is some more information:
> > 
> > [ 2936.169191] wcte12xp0: Missed interrupt. Increasing latency to 6 ms
> > in order to compensate.
> > [ 4734.685566] wcte12xp0: Missed interrupt. Increasing latency to 7 ms
> > in order to compensate.
> > [ 4893.695402] zaptel Disabled echo canceller because of tone (rx) on
> > channel 56
> > [ 5248.845635] wcte12xp: NMF workaround on!
> > [ 5248.845640] wcte12xp: Setting yellow alarm
> > [ 5248.845658] Zaptel: Master changed to XBUS-00/XPD-00
> > [ 5248.845777] wcte12xp0: Missed interrupt. Increasing latency to 8 ms
> > in order to compensate.
> > [ 5248.910831] wcte12xp: NMF workaround off!
> > [ 5253.908078] Zaptel: Master changed to WCT1/0
> > [ 5253.964028] wcte12xp: Clearing yellow alarm
> 
> This should have caused events to be sent on each channel of the span to
> clear the alarm. You should be able to see those events on e.g. the
> "full" log of Asterisk. Do you see them?
> 
> > [ 5335.053981] wcte12xp0: Missed interrupt. Increasing latency to 9 ms
> > in order to compensate.
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] Multiple user registration ...

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mauro Sergio Ferreira Brasil wrote:

> We are planning to use Asterisk on our VoIP platform, and we are 
> spending some brains on a way to provide the following facility: let 
> some SIP user (extension) registrate with more than one client (ATA, 
> SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate 
> calls from any of this devices that are registrated with the same user - 
> no problems on tests too -, but also receive INVITE requests on "all" 
> devices if someone calls this user - yeah... here the thing gets creepy.
> The demand is quite simple: let a user registrate with multiple devices 
> using the same SIP user on such way that if someone call him, all these 
> registered devices will ring and the first to take the call will be "the 
> lucky one".

Instead of trying to make Asterisk do this unnatural act, why not
register each device with a separate id, then use the dial function to
call all of them?

e.g.exten => 122,1,Dial(SIP/1&SIP/2&SIP/3)

You could use some creating scripting to decide which devices to ring
based on the dialed extension.

Barry
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Re: [asterisk-users] Echo

[Jason Baker]

  I'm no echo expert, but if you are hearing your own voice echoed back to 
you on calls the first thing I would check is your txgain settings. 
There's plenty of info on voip-info.org to help you with that.

Thank you, I have messed with that a bit, but I will try again, once I
verify that the hardware echo is working.

I have the VPMADT032 echo cancellation module attached to a Digium
TE121 PCI express card. Does anyone have any experience with the
VPMADT032, or should I be looking for a better solution?


Jason Baker
IT
Coordinator

Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
Fax: 989.752.4276
www.glastender.com



Dave Fullerton wrote:

  Jason Baker wrote:
  
  

  
Echo Cancellation: 128 taps unless TDM bridged, currently ON

The "currently ON" is telling you that the echo canceller is active.

You could try changing echotraining to no in chan_dahdi.conf as well.

What were you running before you upgraded?

  
  So, Asterisk doesn't start echo canceling a line until it is in use? I thought 
that might be the case.

I was running Zaptel before this, not sure what version. I upgrade to Dahdi. The 
echo was present in Zaptel, but not as bad.

Does anyone have any experience with hardware echo cancel modules? Are they 
better/worse than software? What would be the best solution to remove echo?
  

  
  
No, asterisk does not start echo canceling on a channel until the 
channel is brought up.

I have two sites with hardware echo cancellers. One Digium on a TE220B 
and one on a Sangoma A200. I use OLSEC (software) on my home system. 
 From my experience, I would say the hardware solutions work the best, 
though OSLEC is very good.

I'm no echo expert, but if you are hearing your own voice echoed back to 
you on calls the first thing I would check is your txgain settings. 
There's plenty of info on voip-info.org to help you with that.

-Dave

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Re: [asterisk-users] Multiple user registration ...

[Elliot Otchet]

Is your goal here to have multiple devices ring when an extension is dialed and 
the first one to answer take the call?

If so, see the Dial command 
Dial(Technology/resource&Technology/resource&Technology/resource...[|timeout][|options][|URL]).
  When multiple technology/resource entries are listed, the first one to answer 
will take the call.  That accomplishes your goal, if I understand you correctly.

The nice part about doing it this way (with each device independently 
registered) is that you gain a substantial amount of granularity in controlling 
where calls go and you don't have to find creative ways (read: unsupported) to 
trick Asterisk or endpoints.

If you're developing your own GUI to have people set up their devices, you can 
easily create a wizard that walks them through setting up each device and 
associating them together through either channel variables or other tables in a 
database.

I use this methodology in 1.4 and it works quite reliably.  For a good 
reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or 
from your Asterisk console try: 'core show application dial'

It's not perfect because you can have devices that do funny things with a SIP 
INVITE, but in most cases it works very well.

Regards,

Elliot

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio 
Ferreira Brasil
Sent: Wednesday, August 26, 2009 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Multiple user registration ...

Hello there!

We are planning to use Asterisk on our VoIP platform, and we are
spending some brains on a way to provide the following facility: let
some SIP user (extension) registrate with more than one client (ATA,
SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate
calls from any of this devices that are registrated with the same user -
no problems on tests too -, but also receive INVITE requests on "all"
devices if someone calls this user - yeah... here the thing gets creepy.
The demand is quite simple: let a user registrate with multiple devices
using the same SIP user on such way that if someone call him, all these
registered devices will ring and the first to take the call will be "the
lucky one".
The demand, as I've said, is quite simple and logical (translated to our
living world), but the reality is a very different history.

On our tests, always is the last registered application/device that
receives the call indication.
And only the last one.

We are making some tests trying to "kind of deceive" Asterisk on second,
third, and additional, registrations so it receives from Realtime "fake"
extensions numbers on such a way that we can use all these fake
extensions to build a queue dinamicaly (through ARA) and provide the
desired "ring on all" functionality.
I think this will lead us to lots of SIP sinalization and multi user
registration problems, but that was the best shot we had here until now.

I would like to know if anyone had the same demand and, maybe, have
found any viable solution to it.

Thanks and best regards,

--
__At.,
   _

*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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Re: [asterisk-users] application missed in asterisk 1.6.1 - SetCallerID()

[harry R]

> Read the UPGRADE.txt
>
> Solution is to use functions instead:
>
> Set(CALLERID(name));
> Set(CALLERID(num));
> Set(CHANNEL(language));
> etc


Thanks again for solution

Harry.
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Re: [asterisk-users] application missed in asterisk 1.6.1 - SetCallerID()

[Atis Lezdins]

On Wed, Aug 26, 2009 at 5:03 PM, harry R wrote:
> Hi
>
> A few day ago, I notice that some applications missed in asterisk 1.6.1
> release even if *.so file which normally create them were compiled during
> Asterisk install.
> SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so
> more.
>
> anyone already notice that to ?
>
> If it's not normal, anyone have an solution to it ?

Read the UPGRADE.txt

Solution is to use functions instead:

Set(CALLERID(name));
Set(CALLERID(num));
Set(CHANNEL(language));
etc

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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[asterisk-users] Multiple user registration ...

[Mauro Sergio Ferreira Brasil]

Hello there!

We are planning to use Asterisk on our VoIP platform, and we are 
spending some brains on a way to provide the following facility: let 
some SIP user (extension) registrate with more than one client (ATA, 
SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate 
calls from any of this devices that are registrated with the same user - 
no problems on tests too -, but also receive INVITE requests on "all" 
devices if someone calls this user - yeah... here the thing gets creepy.
The demand is quite simple: let a user registrate with multiple devices 
using the same SIP user on such way that if someone call him, all these 
registered devices will ring and the first to take the call will be "the 
lucky one".
The demand, as I've said, is quite simple and logical (translated to our 
living world), but the reality is a very different history.

On our tests, always is the last registered application/device that 
receives the call indication.
And only the last one.

We are making some tests trying to "kind of deceive" Asterisk on second, 
third, and additional, registrations so it receives from Realtime "fake" 
extensions numbers on such a way that we can use all these fake 
extensions to build a queue dinamicaly (through ARA) and provide the 
desired "ring on all" functionality.
I think this will lead us to lots of SIP sinalization and multi user 
registration problems, but that was the best shot we had here until now.

I would like to know if anyone had the same demand and, maybe, have 
found any viable solution to it.

Thanks and best regards,

-- 
__At.,  
   
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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[asterisk-users] application missed in asterisk 1.6.1 - SetCallerID()

[harry R]

Hi

A few day ago, I notice that some applications missed in asterisk 1.6.1
release even if *.so file which normally create them were compiled during
Asterisk install.
SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so
more.

anyone already notice that to ?

If it's not normal, anyone have an solution to it ?
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Re: [asterisk-users] Realtime with "rtcachefriends=no" problems...

[Mauro Sergio Ferreira Brasil]

Thanks Atis, its working pretty fine now.

Best regards,
Mauro.



Atis Lezdins escreveu:
> On Wed, Aug 26, 2009 at 12:11 AM, Mauro Sergio Ferreira
> Brasil wrote:
>   
>> Hello there!
>>
>> Problem found.
>>
>> For some reason, the update statement below is generated with an invalid
>> atribution of empty value '' to field "port" that is an integer.
>> Because of that, this record keeps with prior "fullcontact" information
>> that was updated by another client (which uses a different port) what
>> leads to wrong client rtp packets routing... wow... that was weird... :-)
>>
>> [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime:
>> Query: UPDATE sip_buddies SET fullcontact = '', ipaddr = '', port = '',
>> regseconds = '0', username = '', regserver = '' WHERE name = '101'
>> [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query
>> Failed because: Incorrect integer value: '' for column 'port' at row 1
>>
>> First of all... my appologies by the false alarm.
>> But now I need your help to identify why is this update statement being
>> generated wrongly.
>>
>> Does someone have any idea ?
>> 
>
> Asterisk Realtime Architecutre currently treats all fields as strings.
> I wish too that it would take into account actual field type retrieved
> from DESCRIBE statement and add the quotes only if it's string.
>
> You can safely do
>
> ALTER TABLE sip_buddies CHANGE COLUMN port port VARCHAR(5);
>
> Regards,
> Atis
>
>   

-- 
__At.,  
   
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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Re: [asterisk-users] Open Source Visual Call-Flow and IVR Dev Tool v1.0 Released!

[Alex Bell]

The Safisystems Team:This is an awesome app. I'm downloading
it now and will let you know my experiences.

Thanks for all your efforts to the OS Community,
Al

*
*
On Wed, Aug 26, 2009 at 2:48 AM, zac wolfe  wrote:

> After over a year of alphas, betas, and release candidates I'm happy to
> announce that Version 1.0 of SafiServer and SafiWorkshop has just been
> released under the open source license GPL (ver 3). You can download
> installers from our site www.safisystems.com and the source code can be
> downloaded from Sourceforge (more details available on our site).
>
> If you're not familiar with our system and you do any Asterisk voice
> application development at all you're in for a life-altering experience.
> Again, more info is available on our site but here's a quick description:
>
> *SafiWorkshop is a visual call flow designer that allows Asterisk
> administrators to quickly create and deploy powerful IVR, auto-attendants,
> and call routing applications by creating diagrams that reflect the desired
> function. These diagrams, or Saflets, can then be executed remotely on Safi
> Systems’ standalone server component: SafiServer. Key features include full
> featured graphical IVR/logic designer, live remote debugging, built-in audio
> prompt studio for creating, editing and distributing prompts, simple
> application deployment, database connection pooling, multiple Asterisk
> server capable, FreePBX plugin, and many more powerful features that makes
> Asterisk voice application development simple and fun. *
>
> We think this is a great tool as it stands now but your input and
> contributions will help us make it even better -- don't hesitate to get
> involved and post your bugs, suggestions, or code (send a request to
> i...@safisystems.com). We're happy to finally be in a position to give a
> little something back to the OS community that has been so good to us.
> Enjoy!
>
> Zac Wolfe
> Safi Systems LLC
> www.safisystems.com
>
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Re: [asterisk-users] Echo

[Dave Fullerton]

Jason Baker wrote:
>> > Echo Cancellation: 128 taps unless TDM bridged, currently ON
>> >
>> > The "currently ON" is telling you that the echo canceller is active.
>> >
>> > You could try changing echotraining to no in chan_dahdi.conf as well.
>> >
>> > What were you running before you upgraded?
>> So, Asterisk doesn't start echo canceling a line until it is in use? I 
>> thought 
>> that might be the case.
>>
>> I was running Zaptel before this, not sure what version. I upgrade to Dahdi. 
>> The 
>> echo was present in Zaptel, but not as bad.
>>
>> Does anyone have any experience with hardware echo cancel modules? Are they 
>> better/worse than software? What would be the best solution to remove echo?

No, asterisk does not start echo canceling on a channel until the 
channel is brought up.

I have two sites with hardware echo cancellers. One Digium on a TE220B 
and one on a Sangoma A200. I use OLSEC (software) on my home system. 
 From my experience, I would say the hardware solutions work the best, 
though OSLEC is very good.

I'm no echo expert, but if you are hearing your own voice echoed back to 
you on calls the first thing I would check is your txgain settings. 
There's plenty of info on voip-info.org to help you with that.

-Dave

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Re: [asterisk-users] User-invoked call restrictions

[Danny Nicholas]

You could also do this in the dialplan without AGI.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roberto Piola
Sent: Wednesday, August 26, 2009 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] User-invoked call restrictions

Hint: an AGI application that looks into a database for passwords, and
the decides, according to the prefix, if the call is allowed or not

On Wed, Aug 26, 2009 at 4:39 AM, David A. Bandel
wrote:
> Folks,
>
> Had a request from a customer:  is it possible for a customer, using a
> password to restrict others from making long distance/cell calls?
> That is, the user set a level of service?
>
> Something like this:
> Customer dials a number -- "operator" asks for password, then service
> level (another number).  Service level would be something like:
> 1 - allow inbound calls only
> 2 - allow 1 + local/toll free calls
> 3 - allow 2 + long distance national
> 4 - allow 3 + cell calls
> 0 - allow 4 + international calls (basically cancel all call restrictions)
>
> Dialing to restricted zones would evoke a message from the operator
> that the phone is blocked by owner.
>
> Code examples?  Hints?  RTFM URL?
>
> TIA,
>
> David A. Bandel
> --
> Focus on the dream, not the competition.
>            - Nemesis Air Racing Team motto
> Visit my blog at: http://www.pananix.com/cgi-bin/blosxom
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
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Senior Network Engineer
Outsourcing Infrastructure

VISIANT OUTSOURCING

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[asterisk-users] Header from REFFER

[Gomtesh Jain]

Hi All,
   I am using asterisk in my VoIP solution.
   Where one sip service sends "REFFER" with a user defined header  to
asterisk  , but asterisk does not send this header in INVITE to transfered
party .
Is it a bug in asterisk ?
Is there any solution to do this .

Gomtesh
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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Tzafrir Cohen]

On Wed, Aug 26, 2009 at 02:13:13PM +0200, Loic Didelot wrote:

> Does zaptel really need to increase the latency and to change the
> master?

Maybe it does. Maybe it doesn't. How about debugging the actual problem
at hand (alarm was not properly cancelled, according to Asterisk)?

-- 
   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] PRI worked fine for months, now it stopps working after 2-3 hours

[Loic Didelot]

Hi,
finally I got even more logs from dmesg that could be useful.

[ 1380.149049] wcte12xp0: Missed interrupt. Increasing latency to 5 ms
in order to compensate.
[ 1446.621237] wcte12xp: turning off tone detection
[ 1802.551501] wcte12xp: turning off tone detection
[ 1979.801185] wcte12xp0: Missed interrupt. Increasing latency to 6 ms
in order to compensate.
[ 2579.233403] wcte12xp0: Missed interrupt. Increasing latency to 7 ms
in order to compensate.
[ 3179.055821] wcte12xp: NMF workaround on!
[ 3179.055825] wcte12xp: Setting yellow alarm
[ 3179.055847] Zaptel: Master changed to XBUS-00/XPD-00
[ 3179.055941] wcte12xp0: Missed interrupt. Increasing latency to 8 ms
in order to compensate.
[ 3179.120989] wcte12xp: NMF workaround off!
[ 3184.118231] Zaptel: Master changed to WCT1/0
[ 3184.174181] wcte12xp: Clearing yellow alarm
[ 3778.467882] wcte12xp0: Missed interrupt. Increasing latency to 9 ms
in order to compensate.


Does zaptel really need to increase the latency and to change the
master?

Best regards,
Loïc Didelot.


On Wed, 2009-08-26 at 10:34 +0300, Tzafrir Cohen wrote:
> On Wed, Aug 26, 2009 at 08:53:18AM +0200, Loic Didelot wrote:
> > Hello,
> > we have several customers with a PRI line and a Wildcard TE121.
> > Everything worked fine, but now are one customer the PRI stops working
> > after a few hours.
> > 
> > "zap show channel 1" shows that the channel is InAlarm. I the log files
> > of asterisk I see that the D-Channel seems down. Restarting asterisk
> > does not help, but rebooting the whole server resolves the problem for
> > another 2-3 hours.
> 
> The obvious stupid question: Does Zaptel (The kernel) report that the
> span is in alarm? 
> 
> cat /proc/zaptel/1
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] Bria / eyebeam: no RTCP while on hold

[Stanisław Pitucha]

2009/8/26 Paul Herman :
> I use Bria and eyebeam and it seems that asterisk doesn't send RCTP
> keepalives when a SIP channel is on hold.

Slightly related: https://issues.asterisk.org/view.php?id=15466
It also affects integration with OCS for me.

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[asterisk-users] looking for user:Zoa for T38...

[BERGANZ François]

Zoa,

 

I could see that you could do T38 passthrough with asterisk ans Zoiper.

I have some problems to do it, can you help me?

 

 

Cordialement,

P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

 

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Re: [asterisk-users] Echo

[Jason Baker]

  Echo Cancellation: 128 taps unless TDM bridged, currently ON

The "currently ON" is telling you that the echo canceller is active.

You could try changing echotraining to no in chan_dahdi.conf as well.

What were you running before you upgraded?

So, Asterisk doesn't start echo canceling a line until it is in use? I
thought that might be the case.

I was running Zaptel before this, not sure what version. I upgrade to
Dahdi. The echo was present in Zaptel, but not as bad.

Does anyone have any experience with hardware echo cancel modules? Are
they better/worse than software? What would be the best solution to
remove echo?


Jason Baker
IT
Coordinator

Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
Fax: 989.752.4276
www.glastender.com



Dave Fullerton wrote:

  Jason Baker wrote:
  
  
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.

I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
TE121B PCI express card with a HARDWARE echo cancellation module. All
this is housed on a CentOS 5.5 box, 2.6.18 Kernel. My incoming phone
service is an AT&T PRI (24 channel T1).

My configs:

chan_dahdi.conf*

[channels]
; configuration for T1 card as PRI
language = en

group = 1
echocancel = yes
echotraining = yes
signalling = pri_cpe
switchtype = 4ess
usecallerid = yes
context = incoming
channel => 1-23


***/etc/dahdi/system.conf*
loadzone=us
defaultzone=us
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24

When I run dahdi_cfg -vvv I get the following:

DAHDI Tools Version - 2.2.0

DAHDI Version: 2.2.0.1
Echo Canceller(s): MG2
Configuration
==

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01)

  
  
  
  
Channel 23: Clear channel (Default) (Echo Canceler: none) (Slaves: 23)
Channel 24: D-channel (Default) (Echo Canceler: none) (Slaves: 24)

24 channels to configure.

Setting echocan for channel 1 to none

  
  
  
  
Setting echocan for channel 24 to none


It is showing MG2 as the echo canceller, even though I don't have an echo 
canceller specified. Is that the harwdare module? Do I even need to specify an 
echo canceller in the configs if I have a hardware echo module?

  
  
MG2 is a software canceller. I don't think that line means that MG2 is 
being used on all your channels. If you look at the Channel map it says 
"(Echo Canceler: none)". If it had been set to MG2 you would see MG2 
instead of none.

You do not need to specify an echo canceller in system.conf when you 
have a hardware canceller. One thing I would check is to make sure 
asterisk is activating the echo canceller when a call is in progress. To 
do this execute "core show channels" at the asterisk command line (make 
sure someone on the system has placed a call on the PRI). Look for a 
DAHDI/#-x line. Then execute "dahdi show channel #" where # is the 
channel number. You'll get a screen full of output. Look for a line that 
looks like this (it will be near the end):

Echo Cancellation: 128 taps unless TDM bridged, currently ON

The "currently ON" is telling you that the echo canceller is active.

You could try changing echotraining to no in chan_dahdi.conf as well.

What were you running before you upgraded?

-Dave

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[asterisk-users] Bria / eyebeam: no RTCP while on hold

[Paul Herman]

Hi!

I use Bria and eyebeam and it seems that asterisk doesn't send RCTP 
keepalives when a SIP channel is on hold.   This is a known issue as is 
described here:

   http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Bria

This gets very annoying because very often people are put on hold longer 
than 30 seconds (the phone's default.)  In a company with more than 100 
soft phones it is a lot of work to configure each phone for what seems 
like a workaround rather than a solution.

I don't know much about RTP so my question is, anyone know any 
particular reason why asterisk doesn't send RCTP keepalives while on 
hold?  Is this possibly a bug?

Cheers,

Paul Herman

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Re: [asterisk-users] User-invoked call restrictions

[Roberto Piola]

Hint: an AGI application that looks into a database for passwords, and
the decides, according to the prefix, if the call is allowed or not

On Wed, Aug 26, 2009 at 4:39 AM, David A. Bandel wrote:
> Folks,
>
> Had a request from a customer:  is it possible for a customer, using a
> password to restrict others from making long distance/cell calls?
> That is, the user set a level of service?
>
> Something like this:
> Customer dials a number -- "operator" asks for password, then service
> level (another number).  Service level would be something like:
> 1 - allow inbound calls only
> 2 - allow 1 + local/toll free calls
> 3 - allow 2 + long distance national
> 4 - allow 3 + cell calls
> 0 - allow 4 + international calls (basically cancel all call restrictions)
>
> Dialing to restricted zones would evoke a message from the operator
> that the phone is blocked by owner.
>
> Code examples?  Hints?  RTFM URL?
>
> TIA,
>
> David A. Bandel
> --
> Focus on the dream, not the competition.
>            - Nemesis Air Racing Team motto
> Visit my blog at: http://www.pananix.com/cgi-bin/blosxom
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Roberto Piola, Ph.D.
Senior Network Engineer
Outsourcing Infrastructure

VISIANT OUTSOURCING

strada del Drosso 128/6 - 10135 Torino
T +39 011 3473520 - F +39 011 3473522
M +39 3356961505
roberto.pi...@visiant.it

www.visiantoutsourcing.it


Questo messaggio è destinato alle sole persone indicate e può
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Re: [asterisk-users] Follow me IVR sounds

[James Mutuku]

thanks danny for the reply. I am looking into using flite to read out the
prompts. if i ma ask...are there other voices other than the mechanical
robotic male voice available for flite.? I have searched over the internet
and I can't seem to find any
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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Loic Didelot]

Hello,
seems that the IRQ is shared. How can I change this? What would be the
best way to move it to a dedicated IRQ?

Loïc.

On Wed, 2009-08-26 at 12:22 +0200, F6HQZ wrote:
> Hi men,
> 
> And what happens without APIC/ACPI ? I hate them !  Any IRQ sharing issue ?
> 
> Francois
> 
> 
> -Message d'origine-
> From Loic Didelot
> 
> ...SNIP...
> 
> cat /proc/interrupts 
>CPU0   CPU1   
>   0: 83  0   IO-APIC-edge  timer
>   1:  2  0   IO-APIC-edge  i8042
>   3: 100887  0   IO-APIC-edge  serial
>   7:  0  0   IO-APIC-edge  parport0
>   8:  3  0   IO-APIC-edge  rtc
>   9:  1  0   IO-APIC-fasteoi   acpi
>  16:5381227  0   IO-APIC-fasteoi   uhci_hcd:usb3, wcte12xp0,
> eth5
>  17: 570760  0   IO-APIC-fasteoi   eth4
>  18:  0  0   IO-APIC-fasteoi   ehci_hcd:usb1,
> uhci_hcd:usb7
>  19:   20413905  0   IO-APIC-fasteoi   ehci_hcd:usb2,
> uhci_hcd:usb5
>  20:  0  0   IO-APIC-fasteoi   uhci_hcd:usb4
>  21:  33904  0   IO-APIC-fasteoi   uhci_hcd:usb6, libata,
> libata
> NMI:  0  0 
> LOC: 604815 708592 
> ERR:  0
> MIS:  0
> 
> 
> Loic
> Analyse effectuée par AVG - www.avg.fr 
> Version: 8.5.375 / Base de données virale: 270.13.66/2325 - Date: 08/25/09 
> 18:07:00
> 
> 
> 
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-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Tzafrir Cohen]

On Wed, Aug 26, 2009 at 11:53:25AM +0200, Loic Didelot wrote:
> Here is some more information:
> 
> [ 2936.169191] wcte12xp0: Missed interrupt. Increasing latency to 6 ms
> in order to compensate.
> [ 4734.685566] wcte12xp0: Missed interrupt. Increasing latency to 7 ms
> in order to compensate.
> [ 4893.695402] zaptel Disabled echo canceller because of tone (rx) on
> channel 56
> [ 5248.845635] wcte12xp: NMF workaround on!
> [ 5248.845640] wcte12xp: Setting yellow alarm
> [ 5248.845658] Zaptel: Master changed to XBUS-00/XPD-00
> [ 5248.845777] wcte12xp0: Missed interrupt. Increasing latency to 8 ms
> in order to compensate.
> [ 5248.910831] wcte12xp: NMF workaround off!
> [ 5253.908078] Zaptel: Master changed to WCT1/0
> [ 5253.964028] wcte12xp: Clearing yellow alarm

This should have caused events to be sent on each channel of the span to
clear the alarm. You should be able to see those events on e.g. the
"full" log of Asterisk. Do you see them?

> [ 5335.053981] wcte12xp0: Missed interrupt. Increasing latency to 9 ms
> in order to compensate.

-- 
   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] Asterisk Autodialer

[Alex Balashov]

I'm not sure I accurately understood the problem, but it sounds like 
this is a failure to use non-blocking I/O on the client?

Raimund Sacherer wrote:

> oh boy,
> 
> On Aug 25, 2009, at 10:05 PM, Alex Balashov wrote:
> 
>> I've never seen that, myself.  But I have no trouble believing it.
>>
>> That problem - along with Asterisk's other scalability quirks - must  
>> be
>> properly managed.  More boxes to spread the calls onto and
>> underutilising the hardware on each node is a better extreme to tend
>> toward than the opposite.
> 
> I can tell you, it can and will bring down your machine if not used  
> *VERY*, *VERY* carefully!
> 
> I have not once in my life (and i've done a lot of stuff) seen such a  
> dangerous beast as the AMI, if it's an API for a developer, who  
> develops stuff which are intended to be used by implementors, that's  
> fine, they know what they are doing. But the AMI is dangerous like an  
> API, but intended to be used by implementors which not necessarly do  
> have enough background!
> 
> I had to rewrite a Click2Call solution to specifically use Locks to  
> prevent the following situation:
> 
> the callcenter queried from a host of webservers our webserver, which  
> talks via AMI to get the free/occupied info on callagents
> if a customer want to be called this was routed as well through the AMI.
> 
> The problem is that at Times asterisk locks for some seconds the ami  
> interface, at times it is when a call setup takes longer, at times it  
> has nothing to do with call setups at all, it just locked the AMI and  
> the local apache processes which tried to get answers from the AMI  
> kept piling up, so at times the Lock got freed and everything went  
> normal again, but at times the lock took longer, Apache used up all  
> its possible threads (20 worker-servers a 254 connections) and  
> asterisk just locked up completely (100% cpu) load of the machine  
> about 80 or more.
> 
> At times the machine was so completely hosed that you could not even  
> do anything on the local console so you had to cold-reset the machine!
> 
> I now have implemented:
>   * a caching system for occupied requests
>   * a Lock system so I am sure only ONE thread systemwide can speak at  
> a given time with the AMI
>   * changed the call setup from AMI to pbx_spool
> 
> In my oppinion a system integrator should not have to mess around with  
> locks and doing all this debugging and checking, if i have a manager  
> interface it should happily accept whatever i through at it, and do  
> internel locking, checking, discarding on problem totally safe on its  
> own!
> 
> so, just be aware of WHAT you are doing with the AMI!
> 
> best
> Ray
> 
> 
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-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[F6HQZ]

Hi men,

And what happens without APIC/ACPI ? I hate them !  Any IRQ sharing issue ?

Francois


-Message d'origine-
From Loic Didelot

...SNIP...

cat /proc/interrupts 
   CPU0   CPU1   
  0: 83  0   IO-APIC-edge  timer
  1:  2  0   IO-APIC-edge  i8042
  3: 100887  0   IO-APIC-edge  serial
  7:  0  0   IO-APIC-edge  parport0
  8:  3  0   IO-APIC-edge  rtc
  9:  1  0   IO-APIC-fasteoi   acpi
 16:5381227  0   IO-APIC-fasteoi   uhci_hcd:usb3, wcte12xp0,
eth5
 17: 570760  0   IO-APIC-fasteoi   eth4
 18:  0  0   IO-APIC-fasteoi   ehci_hcd:usb1,
uhci_hcd:usb7
 19:   20413905  0   IO-APIC-fasteoi   ehci_hcd:usb2,
uhci_hcd:usb5
 20:  0  0   IO-APIC-fasteoi   uhci_hcd:usb4
 21:  33904  0   IO-APIC-fasteoi   uhci_hcd:usb6, libata,
libata
NMI:  0  0 
LOC: 604815 708592 
ERR:  0
MIS:  0


Loic
Analyse effectuée par AVG - www.avg.fr 
Version: 8.5.375 / Base de données virale: 270.13.66/2325 - Date: 08/25/09 
18:07:00



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Re: [asterisk-users] Realtime with "rtcachefriends=no" problems...

[Atis Lezdins]

On Wed, Aug 26, 2009 at 12:11 AM, Mauro Sergio Ferreira
Brasil wrote:
> Hello there!
>
> Problem found.
>
> For some reason, the update statement below is generated with an invalid
> atribution of empty value '' to field "port" that is an integer.
> Because of that, this record keeps with prior "fullcontact" information
> that was updated by another client (which uses a different port) what
> leads to wrong client rtp packets routing... wow... that was weird... :-)
>
> [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime:
> Query: UPDATE sip_buddies SET fullcontact = '', ipaddr = '', port = '',
> regseconds = '0', username = '', regserver = '' WHERE name = '101'
> [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query
> Failed because: Incorrect integer value: '' for column 'port' at row 1
>
> First of all... my appologies by the false alarm.
> But now I need your help to identify why is this update statement being
> generated wrongly.
>
> Does someone have any idea ?

Asterisk Realtime Architecutre currently treats all fields as strings.
I wish too that it would take into account actual field type retrieved
from DESCRIBE statement and add the quotes only if it's string.

You can safely do

ALTER TABLE sip_buddies CHANGE COLUMN port port VARCHAR(5);

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Loic Didelot]

Here is some more information:

[ 2936.169191] wcte12xp0: Missed interrupt. Increasing latency to 6 ms
in order to compensate.
[ 4734.685566] wcte12xp0: Missed interrupt. Increasing latency to 7 ms
in order to compensate.
[ 4893.695402] zaptel Disabled echo canceller because of tone (rx) on
channel 56
[ 5248.845635] wcte12xp: NMF workaround on!
[ 5248.845640] wcte12xp: Setting yellow alarm
[ 5248.845658] Zaptel: Master changed to XBUS-00/XPD-00
[ 5248.845777] wcte12xp0: Missed interrupt. Increasing latency to 8 ms
in order to compensate.
[ 5248.910831] wcte12xp: NMF workaround off!
[ 5253.908078] Zaptel: Master changed to WCT1/0
[ 5253.964028] wcte12xp: Clearing yellow alarm
[ 5335.053981] wcte12xp0: Missed interrupt. Increasing latency to 9 ms
in order to compensate.

cat /proc/interrupts 
   CPU0   CPU1   
  0: 83  0   IO-APIC-edge  timer
  1:  2  0   IO-APIC-edge  i8042
  3: 100887  0   IO-APIC-edge  serial
  7:  0  0   IO-APIC-edge  parport0
  8:  3  0   IO-APIC-edge  rtc
  9:  1  0   IO-APIC-fasteoi   acpi
 16:5381227  0   IO-APIC-fasteoi   uhci_hcd:usb3, wcte12xp0,
eth5
 17: 570760  0   IO-APIC-fasteoi   eth4
 18:  0  0   IO-APIC-fasteoi   ehci_hcd:usb1,
uhci_hcd:usb7
 19:   20413905  0   IO-APIC-fasteoi   ehci_hcd:usb2,
uhci_hcd:usb5
 20:  0  0   IO-APIC-fasteoi   uhci_hcd:usb4
 21:  33904  0   IO-APIC-fasteoi   uhci_hcd:usb6, libata,
libata
NMI:  0  0 
LOC: 604815 708592 
ERR:  0
MIS:  0


Loic




On Wed, 2009-08-26 at 10:34 +0300, Tzafrir Cohen wrote:
> On Wed, Aug 26, 2009 at 08:53:18AM +0200, Loic Didelot wrote:
> > Hello,
> > we have several customers with a PRI line and a Wildcard TE121.
> > Everything worked fine, but now are one customer the PRI stops working
> > after a few hours.
> > 
> > "zap show channel 1" shows that the channel is InAlarm. I the log files
> > of asterisk I see that the D-Channel seems down. Restarting asterisk
> > does not help, but rebooting the whole server resolves the problem for
> > another 2-3 hours.
> 
> The obvious stupid question: Does Zaptel (The kernel) report that the
> span is in alarm? 
> 
> cat /proc/zaptel/1
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Loic Didelot]

Ok,
the problem just reappeared.


Last lines of dmesg:
[ 2936.169191] wcte12xp0: Missed interrupt. Increasing latency to 6 ms
in order to compensate.
[ 4734.685566] wcte12xp0: Missed interrupt. Increasing latency to 7 ms
in order to compensate.


 cat /proc/zaptel/1
Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER) HDB3/CCS/CRC4 
IRQ misses: 3

   1 WCT1/0/1 Clear (In use) 
   2 WCT1/0/2 Clear (In use) 
   3 WCT1/0/3 Clear (In use) 
   4 WCT1/0/4 Clear (In use) 
   5 WCT1/0/5 Clear (In use) 
   6 WCT1/0/6 Clear (In use) 
   7 WCT1/0/7 Clear (In use) 
   8 WCT1/0/8 Clear (In use) 
   9 WCT1/0/9 Clear (In use) 
  10 WCT1/0/10 Clear (In use) 
  11 WCT1/0/11 Clear (In use) 
  12 WCT1/0/12 Clear (In use) 
  13 WCT1/0/13 Clear (In use) 
  14 WCT1/0/14 Clear (In use) 
  15 WCT1/0/15 Clear (In use) 
  16 WCT1/0/16 HDLCFCS (In use) 
  17 WCT1/0/17 Clear (In use) 
  18 WCT1/0/18 Clear (In use) 
  19 WCT1/0/19 Clear (In use) 
  20 WCT1/0/20 Clear (In use) 
  21 WCT1/0/21 Clear (In use) 
  22 WCT1/0/22 Clear (In use) 
  23 WCT1/0/23 Clear (In use) 
  24 WCT1/0/24 Clear (In use) 
  25 WCT1/0/25 Clear (In use) 
  26 WCT1/0/26 Clear (In use) 
  27 WCT1/0/27 Clear (In use) 
  28 WCT1/0/28 Clear (In use) 
  29 WCT1/0/29 Clear (In use) 
  30 WCT1/0/30 Clear (In use) 
  31 WCT1/0/31 Clear (In use) 


Best regards,
Loïc.


On Wed, 2009-08-26 at 10:34 +0300, Tzafrir Cohen wrote:
> On Wed, Aug 26, 2009 at 08:53:18AM +0200, Loic Didelot wrote:
> > Hello,
> > we have several customers with a PRI line and a Wildcard TE121.
> > Everything worked fine, but now are one customer the PRI stops working
> > after a few hours.
> > 
> > "zap show channel 1" shows that the channel is InAlarm. I the log files
> > of asterisk I see that the D-Channel seems down. Restarting asterisk
> > does not help, but rebooting the whole server resolves the problem for
> > another 2-3 hours.
> 
> The obvious stupid question: Does Zaptel (The kernel) report that the
> span is in alarm? 
> 
> cat /proc/zaptel/1
> 
-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] Asterisk Autodialer

[Raimund Sacherer]

On Aug 26, 2009, at 10:40 AM, Steve Totaro wrote:

>
>
> One of the last things you want to do as the guy who handles the  
> system is do a full reboot when Asterisk becomes completely  
> unresponsive, you have hundreds of agents sitting idle and even more  
> customers calling in.
Amen brother!


This CAN fireback to you a solution provider, big time! Let's face it,  
people are accustomed to nearly 99.99% uptime with telephony, I do not  
understand why asterisk was not designed from day one with this in mind.

Because you CAN design the system that if you have more asterisk  
instances you do not loose existing calls if an asterisk box goes  
down, you could loose

* Calls currently being setup
* Monitored call files (due to possible corruption on crash)
* some seconds in the voice stream (i think loosing a bit of voice and  
having to possibility to say: repeat again weights more than to setup  
the call again)

I knew it is possible out of experience in development, but way more  
because e.g. sipfoundry has a pbx which can do this quite fine, but of  
course it is way less flexible then asterisk.

Currently i am very excited about freeSWITCH, which seems very  
promising in this regard, After reading the philosophy behind  
freeSWITCH from their lead-developer (a long-time asterisk developer)  
i am convinced that, at least for me, freeSWITCH will be more the way  
to go in the future.


ah, so, i am feeling better, a little steam off :-)


>
> Fun times at the next meeting with management!  Monetize your  
> downtime and the AMI can be very expensive.

Yeah, it really is not that pleasant at all :-)

best

-- 
Raimund Sacherer
-
RunSolutions
 Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares


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[asterisk-users] DAHDI PRI ERROR

[DHAVAL INDRODIYA]

hello ALL,

i got following error on Asterisk CLI with DAHDI

my card type is: TE122P
Operating system : centos 5.2
Dahdi Version: DAHDI -linux-2.1.0.4 and dahdi-tools-2.1.0.2
Asterisk Version: 1.6.0.5


* PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 *

actually sometimes this event disconnects call
and it continously displayed on CLI

following are my setting of

/etc/dahdi/system.com

# Global data

loadzone= us
defaultzone = us

span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31

and

 /etc/asterisk/chan_dahdi.conf


[channels]
language=en
context=from-pstn
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
callerid=asreceived
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
resetinterval=never
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
group=0
channel => 1-15
channel => 17-31



I am india based user and Telco provider is TATA


please help me out if any one solved this issue .


regards

Dhaval Indrodiya
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Re: [asterisk-users] Asterisk Autodialer

[Steve Totaro]

On Wed, Aug 26, 2009 at 2:58 AM, Raimund Sacherer wrote:

> oh boy,
>
> On Aug 25, 2009, at 10:05 PM, Alex Balashov wrote:
>
> >
> > I've never seen that, myself.  But I have no trouble believing it.
> >
> > That problem - along with Asterisk's other scalability quirks - must
> > be
> > properly managed.  More boxes to spread the calls onto and
> > underutilising the hardware on each node is a better extreme to tend
> > toward than the opposite.
>
> I can tell you, it can and will bring down your machine if not used
> *VERY*, *VERY* carefully!
>
> I have not once in my life (and i've done a lot of stuff) seen such a
> dangerous beast as the AMI, if it's an API for a developer, who
> develops stuff which are intended to be used by implementors, that's
> fine, they know what they are doing. But the AMI is dangerous like an
> API, but intended to be used by implementors which not necessarly do
> have enough background!
>
> I had to rewrite a Click2Call solution to specifically use Locks to
> prevent the following situation:
>
> the callcenter queried from a host of webservers our webserver, which
> talks via AMI to get the free/occupied info on callagents
> if a customer want to be called this was routed as well through the AMI.
>
> The problem is that at Times asterisk locks for some seconds the ami
> interface, at times it is when a call setup takes longer, at times it
> has nothing to do with call setups at all, it just locked the AMI and
> the local apache processes which tried to get answers from the AMI
> kept piling up, so at times the Lock got freed and everything went
> normal again, but at times the lock took longer, Apache used up all
> its possible threads (20 worker-servers a 254 connections) and
> asterisk just locked up completely (100% cpu) load of the machine
> about 80 or more.
>
> At times the machine was so completely hosed that you could not even
> do anything on the local console so you had to cold-reset the machine!
>
> I now have implemented:
>* a caching system for occupied requests
>* a Lock system so I am sure only ONE thread systemwide can speak at
> a given time with the AMI
>* changed the call setup from AMI to pbx_spool
>
> In my oppinion a system integrator should not have to mess around with
> locks and doing all this debugging and checking, if i have a manager
> interface it should happily accept whatever i through at it, and do
> internel locking, checking, discarding on problem totally safe on its
> own!
>
> so, just be aware of WHAT you are doing with the AMI!
>
> best
> Ray
>

Ray,

Thanks for explaining what I was saying in technical/mechanical terms.

My quote, "The AMI has been notorious for bogging down and halting systems
when used in an intensive way." didn't mean much without the behind the
scenes explanation of the hows and whys.

I just chalked it up to being buggy.

One of the last things you want to do as the guy who handles the system is
do a full reboot when Asterisk becomes completely unresponsive, you have
hundreds of agents sitting idle and even more customers calling in.

Fun times at the next meeting with management!  Monetize your downtime and
the AMI can be very expensive.

-- 
Thanks,
Steve Totaro
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] followme app

[harry R]

   > Hope this helps. Again, bear in mind that we are new to this so if

> > someone suggests a better way, they are probably right :-) - John
>

 Thank you John for this example.
I'll try to implement it and give you a backup if I have any questions or
suggests.

Harry.
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Re: [asterisk-users] app_swift issue

[ABBAS SHAKEEL]

I have tried 0.9.1 1.4.2..1.6  but no success





On Wed, Aug 26, 2009 at 11:33 AM, ABBAS SHAKEEL  wrote:

> Hello
>
> I have installed cepstral  It works woderfull using an agi script but
> .
> when i try to use Swift("say this") is Dial plan  I get the error
>
> [Aug 26 12:30:18] WARNING[7420]: pbx.c:3167 pbx_extension_helper: No
> application 'Swift' for extension (actdemo, 123, 2)
>
>
>
> Now i come to know to install app_swift
>
>
> Here is the issue...
>
> when i try to execute make command on app_swift-1.6.2
>
> I get the following error
>
> [r...@asterisk app_swift-1.6.2]# make
> gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC   -c -o
> app_swift.o app_swift.c
> app_swift.c: In function ‘engine’:
> app_swift.c:402: error: incompatible types in assignment
> app_swift.c: In function ‘load_module’:
> app_swift.c:546: error: ‘AST_MODULE’ undeclared (first use in this
> function)
> app_swift.c:546: error: (Each undeclared identifier is reported only once
> app_swift.c:546: error: for each function it appears in.)
> make: *** [app_swift.o] Error 1
>
>
>
> Now i am thinking to edit app_swift.c but AST_MODULE is not defined in
> app_swift.c
>
> i commented this line ""//#define AST_MODULE "app_swift""
>
> but in vain  Please help
>
> static int load_module(void)
> {
> int res;
> const char *t = NULL;
> struct ast_config *cfg;
> struct ast_flags config_flags = { 0 };
>
> // Set defaults
> cfg_buffer_size = 65535;
> cfg_goto_exten = 0;
> strncpy(cfg_voice, "David-8kHz", sizeof(cfg_voice));
>
> res = ast_register_application(app, engine, synopsis, descrip);
> cfg = ast_config_load(SWIFT_CONFIG_FILE, config_flags);
>
> if (cfg) {
> if ((t = ast_variable_retrieve(cfg, "general", "buffer_size"))) {
> cfg_buffer_size = atoi(t);
> ast_log(LOG_DEBUG, "Config buffer_size is %d\n",
> cfg_buffer_size);
> }
> if ((t = ast_variable_retrieve(cfg, "general", "goto_exten"))) {
> if (!strcmp(t, "yes"))
> cfg_goto_exten = 1;
> else
> cfg_goto_exten = 0;
> ast_log(LOG_DEBUG, "Config goto_exten is %d\n",
> cfg_goto_exten);
> }
>
> ast_config_destroy(cfg);
>
> } else {
> ast_log(LOG_NOTICE, "Failed to load config\n");
> }
>
> return res;
> }
>
> char *description(void)
> {
> return tdesc;
> }
>
>
>
> #define AST_MODULE "app_swift"
>
> AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Cepstral Swift TTS
> Application");
>
>
>
>
>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>


-- 
Best Regards
Shakeel Abbas
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Re: [asterisk-users] PRI worked fine for months, now it stopps working after 2-3 hours

[Tzafrir Cohen]

On Wed, Aug 26, 2009 at 08:53:18AM +0200, Loic Didelot wrote:
> Hello,
> we have several customers with a PRI line and a Wildcard TE121.
> Everything worked fine, but now are one customer the PRI stops working
> after a few hours.
> 
> "zap show channel 1" shows that the channel is InAlarm. I the log files
> of asterisk I see that the D-Channel seems down. Restarting asterisk
> does not help, but rebooting the whole server resolves the problem for
> another 2-3 hours.

The obvious stupid question: Does Zaptel (The kernel) report that the
span is in alarm? 

cat /proc/zaptel/1

-- 
   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] Asterisk 1.6 t 38 passthrough

[BERGANZ François]

Some news:
Fax-asterisk-gatewayt38

Fax->* invite(g711)
...
*->faxringing+200OK
Fax->* invite(T38)
* accept the T38 and reply trying
*->* invite (t38)
* find chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate 
host or port in SDP. Expect audio problems


The problem is that asterisk generate an invite (t38) for himself with others 
parameters in the SDP


Why?



Cordialement,
 Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

-Message d'origine-
De : asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming
Envoyé : vendredi 21 août 2009 17:11
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough

BERGANZ François wrote:
> I have that problem:
> 
> [Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp: 
> Failed to read an alternate host or port in SDP. Expect audio problems
> [Aug 21 15:57:35] WARNING[5198]: chan_sip.c:17425 handle_request_invite: 
> Failed to set an alternate media source on glared reinvite. Audio may not 
> work properly on this call.
> [Aug 21 15:57:37] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp: 
> Failed to read an alternate host or port in SDP. Expect audio problems
> [Aug 21 15:57:37] WARNING[5198]: chan_sip.c:17425 handle_request_invite: 
> Failed to set an alternate media source on glared reinvite. Audio may not 
> work properly on this call.
> 
> Why?
> It is at the second invite to do T38

That would mean that the second INVITE happened at an improper time;
please open an issue on issues.asterisk.org, and include a complete
console log include 'core set verbose 10', 'core set debug 10' and 'sip
set debug on'.

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


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Re: [asterisk-users] Asterisk Autodialer

[Raimund Sacherer]

oh boy,

On Aug 25, 2009, at 10:05 PM, Alex Balashov wrote:

>
> I've never seen that, myself.  But I have no trouble believing it.
>
> That problem - along with Asterisk's other scalability quirks - must  
> be
> properly managed.  More boxes to spread the calls onto and
> underutilising the hardware on each node is a better extreme to tend
> toward than the opposite.

I can tell you, it can and will bring down your machine if not used  
*VERY*, *VERY* carefully!

I have not once in my life (and i've done a lot of stuff) seen such a  
dangerous beast as the AMI, if it's an API for a developer, who  
develops stuff which are intended to be used by implementors, that's  
fine, they know what they are doing. But the AMI is dangerous like an  
API, but intended to be used by implementors which not necessarly do  
have enough background!

I had to rewrite a Click2Call solution to specifically use Locks to  
prevent the following situation:

the callcenter queried from a host of webservers our webserver, which  
talks via AMI to get the free/occupied info on callagents
if a customer want to be called this was routed as well through the AMI.

The problem is that at Times asterisk locks for some seconds the ami  
interface, at times it is when a call setup takes longer, at times it  
has nothing to do with call setups at all, it just locked the AMI and  
the local apache processes which tried to get answers from the AMI  
kept piling up, so at times the Lock got freed and everything went  
normal again, but at times the lock took longer, Apache used up all  
its possible threads (20 worker-servers a 254 connections) and  
asterisk just locked up completely (100% cpu) load of the machine  
about 80 or more.

At times the machine was so completely hosed that you could not even  
do anything on the local console so you had to cold-reset the machine!

I now have implemented:
* a caching system for occupied requests
* a Lock system so I am sure only ONE thread systemwide can speak at  
a given time with the AMI
* changed the call setup from AMI to pbx_spool

In my oppinion a system integrator should not have to mess around with  
locks and doing all this debugging and checking, if i have a manager  
interface it should happily accept whatever i through at it, and do  
internel locking, checking, discarding on problem totally safe on its  
own!

so, just be aware of WHAT you are doing with the AMI!

best
Ray


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