asterisk users - Nov 2003 - Unable to create channel of type 'SIP'

I recently moved my Asterisk configuration to a new server and re-built
Asterisk from CVS. Now, I'm experiencing the following issue with SIP:

Executing Dial("Zap/1-1", "SIP/100|20") in new stack
NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to
create channel of type 'SIP'
  == Everyone is busy at this time

Has anyone seen this issue before?

Thanks.

jeff.gunther@intalgent.com wrote:
>I recently moved my Asterisk configuration to a new server and re-built
>Asterisk from CVS. Now, I'm experiencing the following issue with SIP:
>
>Executing Dial("Zap/1-1", "SIP/100|20") in new stack
>NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to
>create channel of type 'SIP'
>  == Everyone is busy at this time
>
>Has anyone seen this issue before?
>  
>
cvs update again


Jeremy McNamara

Hi
I get the following error when i dial a sip extension, please help 

NOTICE[1681]: app_dial.c:746 dial_exec: Unable to create channel of type
'SIP'
  == Everyone is busy/congested at this time
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I (601) call one of my users (8862), after one minute I try to call him 
again  and get "Unable to create channel of type 'SIP' "

sip show peers does not list him.

I cannot figure out why this happens and more important how I can fix it.



   -- Executing Dial("SIP/601-0f22", "SIP/8862|60|tr") in
new stack
Apr 21 15:30:47 NOTICE[14706]: app_dial.c:973 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing NoOp("SIP/601-0f22", "CONGESTION") in new
stack
    -- Executing VoiceMail("SIP/601-0f22", "u8862") in new
stack
    -- Playing '/var/spool/asterisk/voicemail/others/8862/unavail' 
(language 'en')
    -- Playing 'vm-intro' (language 'en')
  == Spawn extension (VoIP_Phone, 8862, 3) exited non-zero on
'SIP/601-0f22'


bye

Ronald

Hi, i have a working asterisk version CVS-v1-0-05/13/05-15:06:32, i was 
installed using amportal , i want to migrate to another server, this 
time i dont wat to use amportal and edited "by hand" everyfile, i can 
make outboundcalls without problems, but i cant receive anything, either 
from between the sip phones or the external peers, i copied the sip.conf 
from the old server, this is the relevant port of the external peer, a 
cisco as5400:


### sip.conf ###
[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=g723.1
allow=alaw
allow=ulaw
context = bogon-calls ; Send unknown SIP callers to this context
callerid = Unknown
language=es

register => @prepago-in

[prepago-in]
type=friend
host=aaa.bbb.ccc.ddd
context = from-external
dtmfmode=rfc2833
insecure=very ; required for incoming FWD calls

[prepago-out]
type=peer                      ; we only want to call out, not be called
host=aaa.bbb.ccc.ddd
dtmfmode=rfc2833

[22662124]
callerid="22662124" <22662124>
context=from-internal
host=dynamic
secret=22662124
type=friend
username=22662124



this is the error log

Destroying call '4110b0c75ff4e8f5217ccd7703010157@200.13.161.27'
Feb 23 19:16:52 NOTICE[1023]: app_dial.c:1011 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3 - No route to destination)
Transmitting (no NAT) to aaa.bbb.ccc.ddd:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP  aaa.bbb.ccc.ddd:5060;received=aaa.bbb.ccc.ddd
From: <sip:22660124@aaa.bbb.ccc.ddd>;tag=1FA0C538-A3B
To: <sip:22662124@aaa.bbb.ccc.ddd>;tag=as3aaf6cd5
Call-ID: A3C0864C-A40811DA-876CF280-F55453B3@aaa.bbb.ccc.ddd
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:22662124@aaa.bbb.ccc.ddd>
Content-Length: 0
X-Asterisk-HangupCause: No route to destination

what can we wrong?
---
Miguel