SIPGATE Basic with Asterisk 13.9.1 ? and artelis.de?

Hello,
i tyed the Config from Sipgate ***, but did not get it running.
No outgoing and incomeing calls.
But i can see at sipgate.de a connected Phone.

*** https://www.sipgate.de/basic/konfiguration/219/asterisk-pbx

Tyed some diffrent Configs too.

Did someone use a German Sipgate Account (Basic) with Asterisk 13.9.x ?
And can share his sip.conf and extensions.conf.

If got a Account from my DSL-Provider, the SIP Provider is: artelis.de
did anyone has this running ?

My Target is: get calls from: sipgate.de and artelis.de
outgoing call only from: artelis.de

Asterisk is running on the DSL-Router.

Thanks
Dennis

lede*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
sipgate-MySipgateID/MySipgateID 217.xx.xx.x Auto (No) No 5060 OK (31 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

lede*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sipgate.de:5060 N MySipgateID 105 Registered Sun, 19 Feb 2017 15:55:46
1 SIP registrations.

sip.conf

[global]
disallow=all
allow=g722
allow=alaw
allow=ulaw

[sipgate-MySipgateID]
type=peer
insecure=invite
defaultuser=MySipgateID
fromuser=MySipgateID
fromdomain=sipgate.de
secret=MyPassword
host=sipgate.de
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context=sipgate-in
callbackextension=MySipgateID
outboundproxy=sipgate.de
trustrpid=yes
sendrpid=no

extensions.conf

[sipgate-in]
exten => MySipgateID,1,Goto(tel1_in,s,1)

[tel1_out]
exten => _Z,1,Goto(out_internal,${EXTEN},1)
exten => _[+0-9].,1,Gosub(out_sipgate,${EXTEN})

[tel2_out]
exten => _Z,1,Goto(out_internal,${EXTEN},1)
exten => _[+0-9].,1,Gosub(out_sipgate,${EXTEN})

[ltq1_out]
;exten => _[+0-9]!,1,Goto(tel1_out,${EXTEN},1)
exten => _[+0-9]!,1,Dial(local/${EXTEN}@tel1_out/n)

[ltq2_out]
;exten => _[+0-9]!,1,Goto(tel2_out,${EXTEN},1)
exten => _[+0-9]!,1,Dial(local/${EXTEN}@tel2_out/n)

[ltq1_in]
exten => s,1,Dial(TAPI/1,30,t)

[sipgate-out]
exten => _[+0-9].,1,Set(CALLERID(num)=NUMBER)
exten => _[+0-9].,2,Dial(SIP/sipgate/${EXTEN},30,Trg)
exten => _[+0-9].,3,Hangup

If i call my Sipgate Number got this Message in: /var/log/asterisk/messages

[Feb 19 16:01:22] NOTICE[3344] chan_sip.c: Peer ‘sipgate-MySipgateID’ is now Reachable. (30ms / 2000ms)

[Feb 19 16:01:57] WARNING[3351][C-00000000] pbx.c: Channel ‘SIP/sipgate-MySipgateID-00000000’ sent to invalid extension but no invalid handler: context,exten,priority=tel1_in,s,1

You don’t have a tel1_in context, but you are trying to direct calls to it. Not specifically a sipgate issue.

Nor do you have an outsipgate context.

thanks david,

i tyed this config too: https://forum.openwrt.org/viewtopic.php?id=62696

but did not work for me too.

sip.conf

[global]

register => MySigateID:MyPassword@sipgate.de/MySigateID

[sipgate]
type=peer
host=sipgate.de
fromdomain=sipgate.de
dtmfmode=rfc2833
insecure=port,invite
directmedia=no
transport=udp,tcp
context=in_sipgate
defaultuser=MySigateID
fromuser=MySigateID
secret=MyPassword
qualify=yes

extensions.conf

[out_sipgate]
exten => _[+0-9].,1,Set(CALLERID(num)=NUMBER)
exten => _[+0-9].,2,Dial(SIP/sipgate/${EXTEN},30,Trg)
exten => _[+0-9].,3,Hangup

[in_sipgate]
exten => MySipgateID,1,Goto(tel1_in,s,1)

[tel1_out]
exten => _Z,1,Goto(out_internal,${EXTEN},1)
exten => _[+0-9].,1,Gosub(out_sipgate,${EXTEN})

[tel2_out]
exten => _Z,1,Goto(out_internal,${EXTEN},1)
exten => _[+0-9].,1,Gosub(out_sipgate,${EXTEN})

[ltq1_out]
;exten => _[+0-9]!,1,Goto(tel1_out,${EXTEN},1)
exten => _[+0-9]!,1,Dial(local/${EXTEN}@tel1_out/n)

[ltq2_out]
;exten => _[+0-9]!,1,Goto(tel2_out,${EXTEN},1)
exten => _[+0-9]!,1,Dial(local/${EXTEN}@tel2_out/n)

[ltq1_in]
exten => s,1,Dial(TAPI/1,30,t)

[ltq2_in]
exten => s,1,Dial(TAPI/2,30,t)

[tel1_in]
;exten => s,1,Goto(ltq1_in,s,1)
exten => s,1,Dial(local/s@ltq1_in/n)

[tel2_in]
;exten => s,1,Goto(ltq2_in,s,1)
exten => s,1,Dial(local/s@ltq2_in/n)

[lantiq1]
include => ltq1_out

[lantiq2]
include => ltq2_out

cat /var/log/asterisk/messages
[Feb 19 16:23:22] Asterisk 13.9.1 built by dennis @ localhost on a x86_64 running Linux on 2017-02-08 05:52:42 UTC
[Feb 19 16:23:22] NOTICE[3612] cdr.c: CDR simple logging enabled.
[Feb 19 16:23:22] NOTICE[3612] loader.c: 45 modules will be loaded.
[Feb 19 16:23:22] WARNING[3612] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Feb 19 16:23:22] WARNING[3612] chan_sip.c: Section ‘global’ lacks type
[Feb 19 16:23:22] WARNING[3612] chan_sip.c: ‘tcp’ is not a valid transport type when tcpenable=no. If no other is specified, the defaults from general will be used.
[Feb 19 16:23:23] NOTICE[3612] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Feb 19 16:23:23] NOTICE[3637] chan_sip.c: Peer ‘sipgate’ is now Reachable. (30ms / 2000ms)

Contains exactly the same errors that I have already pointed out.

In addition you should fix the, self explanatory, warnings about faulty configuration files.

thanks david,

Contains exactly the same errors that I have already pointed out.

can you give me a helping hand with that ?

i chaned the sip.conf

[global]
directmedia=no
nat=no
type=peer

register => MySipgateID:MyPassword@sipgate.de/MySipgateID

[sipgate]
type=peer
host=sipgate.de
fromdomain=sipgate.de
dtmfmode=rfc2833
insecure=port,invite
context=in_sipgate
defaultuser=MySipgateID
fromuser=MySipgateID
secret=MyPassword

cat /var/log/asterisk/messages
[Feb 19 16:39:04] Asterisk 13.9.1 built by dennis @ localhost on a x86_64 running Linux on 2017-02-08 05:52:42 UTC
[Feb 19 16:39:04] NOTICE[3888] cdr.c: CDR simple logging enabled.
[Feb 19 16:39:04] NOTICE[3888] loader.c: 45 modules will be loaded.
[Feb 19 16:39:04] WARNING[3888] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Feb 19 16:39:04] NOTICE[3888] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge

But now no i can see this:

lede*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
global (Unspecified) No No 0 Unmonitored
sipgate/MySipgateID 217.xx.xx.x Auto (No) No 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline]

There were no fatal errors in your sip.conf. Fix the errors I pointed out in the dialplan first.

The errors should be obvious to anyone who has a sufficient understanding to configure Asterisk, and this is not the place for a full introductory course.

Try somewhere around here: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-1.html#asterisk-DP-Basics-SECT-1.1