RE: [asterisk-users] International Carriers

2007-01-27 Thread Juan Miguel Yamakawa 
Hello Facundo, i have an entreprise in Peru, if you want i can give you a
best price for call in peru.
My traffic is on net.


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Facundo
Ameal
Enviado el: Viernes, 26 de Enero de 2007 10:55 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] International Carriers


Hello everyone!
I 've looking for carriers which can terminate my international calls.
They must accept payments from Argentina and give me interconection to
my Asterisk. I'd appreciate your help or recomendations.


Regards.

--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088

Share your knowledge, use free software.
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[asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or address (6)

2006-09-15 Thread Juan Miguel Yamakawa 



Help me please..

ZT_SPANCONFIG failed on span 1: No such device or 
address (6)

how can i fixed this problem.

Thank you.

JmiguelY
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Re: [Asterisk-Users] Asterisk In ternal sip calls I can´t send/recive

2006-05-29 Thread Juan Miguel Yamakawa 

Hola Omar:

solo cambia tu extension.conf

[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Dial(SIP/200,60,Ttr)
exten = s,5,Dial(SIP/201,60,Ttr)
exten = s,6,Dial(SIP/202,60,Ttr)
exten = s,7,Dial(SIP/203,60,Ttr)


Saludos.


- Original Message - 
From: Omar Lopez Limonta [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 12:46 PM
Subject: [Asterisk-Users] Asterisk Internal sip calls I can´t send/recive



When i made internal call into my LAN using x-lite sip phone client I
retrive in askterisk CLI :

---
ERROR
--
Verbosity is at least 6
   -- Remote UNIX connection
   -- Executing Dial(SIP/201-979d, SIP/201|60|t) in new stack
   -- Called 201
May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
 == No one is available to answer at this time
   -- Executing VoiceMail(SIP/201-979d, 201) in new stack
   -- Playing 'vm-intro' (language 'es')
 == Spawn extension (anurix, 201, 2) exited non-zero on 'SIP/201-979d'
May 29 18:09:34 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 52991
(Non-critical Response)
-

(192.168.1.44 is the Asterisk HOST)

I can do outgoing calls with Zap interface without problems, only i
__can´T__ do calls into my lan with SIP phone/protocol  , i can listen
voicemail because is the second action on extesion.

These are my configuration files:

sip.conf
-

[203]
type=friend
qualify=yes
username=203
secret=203
host=dynamic
callerid=\JuanI\ 203
canreinvite=no
reinvite=no
context = anurix
transfer=yes
mailbox=203
callgroup=1
pickupgroup=1
nat=never
--
extensions.conf
--
[exterior]
exten = _0.,1,Dial(Zap/1/${EXTEN:1},60,r)
exten = _0.,2,Hangup
;Contestar llamada
[entrada]
exten = s,1,Wait,11
exten = s,2,Answer
exten = s,3,Wait,1
exten = 1,1,Dial(SIP/200,60,Ttr)
exten = 2,1,Dial(SIP/201,60,Ttr)
exten = 3,1,Dial(SIP/202,60,Ttr)
exten = 4,1,Dial(SIP/203,60,Ttr)

;BUZONES DE VOZ DESAHABILITADOS

[anurix]
include = exterior
exten = 200,1,Dial(SIP/200,60,t)
exten = 200,2,Voicemail(200)
exten = 200,3,Hangup
exten = 201,1,Dial(SIP/201,60,t)
exten = 201,2,Voicemail(201)
exten = 201,3,Hangup
exten = 202,1,Dial(SIP/202,60,t)
exten = 202,2,Voicemail(202)
exten = 202,3,Hangup
exten = 203,1,Dial(SIP/203,60,t)
exten = 203,2,Voicemail(203)
exten = 203,3,Hangup


--
http://www.tuactualidad.com
IM: pollo.es.pollo en gmail.com
Te lo traigo fresco.








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Re: Re: [Asterisk-Users] doubts about asterisk configuration fromdatabase

2006-05-28 Thread Juan Miguel Yamakawa 

Hello:

Do you need install Mysql-devel.

Best Regards

- Original Message - 
From: 吴应芳 [EMAIL PROTECTED]

To: Asterisk Users Mailing List - No asterisk-users@lists.digium.com
Sent: Monday, May 29, 2006 12:04 AM
Subject: Re: Re: [Asterisk-Users] doubts about asterisk configuration 
fromdatabase




another questions！
According  asterisk realtime sip webpage，I had done following steps：

(1) Make, make install asterisk-addons then copy res_mysql.conf.sample to 
res_mysql.conf and edit the res_mysql.conf with my databases parameter

(2) Edit extconfig.conf ---add
sip.conf = mysql,asterisk,sipfriends
(3) Create a sipfriends table in asterisk database and register some sip 
phone into the table

(4) then restart asterisk...

but ...

*CLI May 29 12:50:08 WARNING[22360]: config.c:920 find_engine: Realtime 
mapping for 'sippeers' found to engine 'mysql', but the engine is not 
available


ask: MYSQL given database driver would be ?


thanks~~





hello,,

Yes, asterisk can use realtime mode



On 5/29/06, 吴应芳 [EMAIL PROTECTED] wrote:

hi,
 I want to complete asterisk configuration from database(MYSQL),now I 
come across some doubts:
1. 
http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+databasediff2=3
says that   Dynamic 'friends' (Asterisk v1.0.*) and the number of 
options supported by this 'MySQL_Friends' system is currently very 
limited,at the same time I find   asterisk-1.2.* don't provide this 
functions,why?  for other factors?
2. If I want to do it with asterisk 1.2.*, do I need to add only 
chan_sip.c and Makefile files in channels directory?
3.Is there any other way to complete asterisk configuration from 
database?


thanks !



BEST REGARDS!

Sharon


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--
Jeffery

　　`∧ ∧��
　　ミ^r^ミ灬)~


iaxtel Num: 1-700-576-1311
fwdnet Num: 728150
http://www.diaip.com
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