Omar Lopez Limonta
2006-May-29 10:46 UTC
[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 0afcadc1422e63800115943201a885fb@192.168.1.44
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
C15A57EC-51A0-4157-BCE5-B09C0A99FD26@192.168.1.33 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.
Juan Miguel Yamakawa
2006-May-29 11:45 UTC
Re: [Asterisk-Users] Asterisk Internal sip calls I can´t send/recive
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" <pollo.es.pollo@gmail.com> 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 0afcadc1422e63800115943201a885fb@192.168.1.44 > 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 > C15A57EC-51A0-4157-BCE5-B09C0A99FD26@192.168.1.33 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. >--------------------------------------------------------------------------------> _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Omar Lopez Limonta
2006-May-29 12:07 UTC
Re: [Asterisk-Users] Asterisk Internal sip calls I can´t send/recive
On 5/29/06, Juan Miguel Yamakawa <jmiguely@gmail.com> wrote:> 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) >Gr?cias Juan probar? ma?ana en el trabajo :D -- http://www.tuactualidad.com IM: pollo.es.pollo en gmail.com Te lo traigo fresco.