[Asterisk-Users] sipgate cannot dial out / loop detected

[Christoph Kampka]

Hi all,
I have troubles getting outgoing calls to work with sipgate. Tried all
possible constellations, copied posted configs, nothing. Registration or
incoming calls never a problem. If I use a SIP phone from my network it
works. * doesn't.

I always get 482 Loop Detected
--
-- Starting simple switch on 'Zap/2-1'
-- Executing SetCallerID("Zap/2-1", ""8004277" <8004277>") in new stack
-- Executing Dial("Zap/2-1", "SIP/$(EXTEN:3)@sipgate") in new stack
-- Called $(EXTEN:3)@sipgate
-- Got SIP response 482 "Loop Detected" back from 217.10.79.9
-- Now forwarding Zap/2-1 to 'Local/$(EXTEN:3)@sipincoming' (thanks to
SIP/sipgate-92b0)
Oct 16 12:36:29 NOTICE[278554]: chan_local.c:374 local_alloc: No such
extension/context [EMAIL PROTECTED](EXTEN:3) creating local channel
Oct 16 12:36:29 NOTICE[278554]: app_dial.c:232 wait_for_answer: Unable to
create local channel for call forward to 'Local/$(EXTEN:3)@sipincoming'
  == Everyone is busy/congested at this time
-- Executing Hangup("Zap/2-1", "") in new stack
  == Spawn extension (default, 9201, 3) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'

Below my current configs. What am I doing wrong?


---
sip.conf

[general]
port=5060
bindaddr=192.168.1.9
externip=xx.dyndns.org
localnet=192.168.1.0/255.255.255.0
context=sipincoming
srvlookup=yes
tos=reliability
disallow=all
allow=ulaw
allow=alaw
allow=gsm

register => 8004277:[EMAIL PROTECTED]/8004277

[sipgate]
type=friend
username=8004277
secret=xx
host=sipgate.de
insecure=very
nat=yes
canreinvite=no
fromuser=8004277
fromdomain=sipgate.net


-
extensions.conf

exten => _920.,1,SetCallerID("8004277" <8004277>)
exten => _920.,2,Dial(SIP/$(EXTEN:3)@sipgate)
exten => _920.,3,Hangup
---
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[Asterisk-Users] Codecs Problem?

[Christoph Kampka]

Hello,
I have a following setup:

IP phone (Cisco/Skinny) <-> * <-> NAT -- NAT <-> * <-> PSTN

Everything is perfect when i'm using it from right to left. From left to
right however, there is no voice, although the calls are being placed.

I played around with codeces but no change.

Does anybody know, what I possibly am doing wrong?

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[Asterisk-Users] Cisco 30 VIP

[Christoph Kampka]

Hi there,
I know this was discussed many times already, but after trying to get my
30VIP to work for a week, I think I might be doing something wrong.
I'm trying to set it up with chan_skinny. Haven't tried chan_sccp yet, but
since there are many of you already doing it with skinny, it should be
working also in my case. I read and used all the hints and samples I was
able to find on the list and elsewhere. Source is new, from yesterday.

This is in my skinny.conf (apart from all the default stuff)

[30vip]
device=SEP00B06409B25D
version=P002F202
context=default
line => 100

I tried also other parameters, to no avail.
Display shows F2.02

Phone settings:
IP 192.168.1.3
Mask 255.255.255.0
Gateway 192.168.1.9 (* box)
DNS 0.0.0.0 (disabled)
TFTP 192.168.1.9 (* box)

Any suggestions?
4

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