Can't register sip hard phones on 1.2.1 but register 1.0.9


#1

a) asterisk 1.2.1 compiles ok installation ok (except mpg123) but can’t register sip hard phones on 1.2.1 suse9.3 x86 but register on 1.0.9 suse9.3 32bit

  • config files are the same
    b) can 1.0.9 run on a PentiumD 64bit pc and benefit at least dual proc power?

#2

What error messages are you getting when you try to register a phone on 1.2.1?


#3

verbosing at 3 level no errors displayed


#4

try typing this in the asterisk console and paste the output here:
sip show settings


#5

Global Settings:

SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm telc
Realm. auth: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS: 0x0
OSP Support: No
SIP realtime: Disabled
telc*CLI>
Global Signalling Settings:

Codecs: gsm,ulaw,alaw
Relax DTMF: No
Compact SIP headers: No
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes

Default Settings:

Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: gk
Musicclass: default
Voice Mail Extension: asterisk

and when dialing …

*CLI> dial 96932437…
– Executing Dial(“OSS/dsp”, “capi/g1/6932437…”) in new stack
– Called g1/6932437…
> CAPI INFO 0x3301: Protocol error layer 1 (broken line or B-channel removed by signalling protocol)
== ISDN1: CAPI Hangingup
== No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘NOANSWER’
<< Hangup on console >>


#6

dear stoffell
please help


#7

Weird, your sip settings look correct. Can you try using an ethernet sniffer to determine if your sip connection is even tried? (to make sure it reaches the server correctly)
Perhaps you can post your sip.conf and extensions.conf?


#8

with * 1.2.2 fixed but

  1. can’t hear calls from one hard phone to another
  2. can’t dial out
  3. can’t dial in

please see

forums.digium.com/viewtopic.php? … ab425d75b8