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Telekom SIP Trunk


(Michael Fuchs) #1

Hi there,
has anybody a working chan_sip or chan_pjsip config he could share with me?
My problem is, that I can receive calls from the outside via the Telekom SIP trunk, but I can not place any calls probably due to a failing registration…
Cheers Michael


(Avayax) #2

Best is you share your config and post logs of a failing call.


(Michael Fuchs) #3

that was quick :wink:
here it is:
I followed the instructions mentioned here, and I succeeded in being able to dial internal phones and reach them from the outside, but when I try to dial out, that is when the problems start…
My first guess is, that it is a registration problem. When I do a “sip show peers” I get
idefix*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
13/13 192.168.12.79 D No No A 5060 OK (23 ms)
14 (Unspecified) D No No A 0 UNKNOWN
15/15 192.168.12.56 D No No A 5060 OK (27 ms)
16 (Unspecified) D No No A 0 UNKNOWN
21 (Unspecified) D No No A 0 UNKNOWN
551133016564/551133016564 217.0.15.67 Yes Yes 5060 OK (19 ms)
TelekomSIPTrunk4Testing/5 (Unspecified) Yes Yes 0 UNKNOWN
7 sip peers [Monitored: 3 online, 4 offline Unmonitored: 0 online, 0 offline]

I do not understand, why it shows 2 peers for the telekom, the one that did not register is “TelekomSIPTrunk4Testing”, which is the trunk name in my outgoing chan_sip config.

Here is my chan_sip config:

Trunk Name TelekomSIPTrunk4Testing

PEER Details
videosupport=no
usereqphone=yes
type=peer
transport=tcp
srvlookup=yes
session-timers=refuse
secret=XXXXXX
qualify=yes
insecure=invite
host=reg.sip-trunk.telekom.de
fromdomain=sip-trunk.telekom.de
dtmfmode=rfc2833
disallow=all
directmedia=no
defaultuser=55YYYYYYYYY
allow=g722,alaw

User Context 55YYYYYYYY
User Details secret=XXXXXXXX
type=peer
transport=tcp
srvlookup=yes
insecure=invite
host=reg.sip-trunk.telekom.de
fromdomain=sip-trunk.telekom.de
defaultuser=55YYYYYYYYY
disallow=all
allow=g722,alaw
qualify=yes
context=from-pstn-toheader

tcp://"+49ZZZZZZZZZZ"@“sip-trunk.telekom.de”:“XXXXXXX”:“55YYYYYYYYYY”@reg.sip-trunk.telekom.de:5060

On the cli it says:

[2019-03-17 17:44:13] ERROR[1528]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo(“reg.sip-trunk.telekom.de”, “(null)”, …): Name or service not known
[2019-03-17 17:44:13] WARNING[1528]: acl.c:835 resolve_first: Unable to lookup ‘reg.sip-trunk.telekom.d(e)’

And when I to dial out:

-- Executing [s@macro-dialout-trunk:24] Dial("SIP/15-00000035", "SIP/TelekomSIPTrunk4Testing/064130020551,300,T") in new stack

[2019-03-17 17:46:25] WARNING[8245][C-0000002f]: app_dial.c:2527 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-dialout-trunk:25] NoOp(“SIP/15-00000035”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20”) in new stack

BTW, it is an asterisk 13 with a FreePBX 13…

Cheers Michael


(Trunk) #4

I’m also using a Telekom sip trunk. I’m using pjsip and everything is working properly. I was trying chan_sip before but it didn’t work correctly. This is my configuration:

General

Username: xxxxxxxxxxxx
Kennwort: yyyyyyyy
Authentication: Outbound
Registration: Send
Language Code: German
SIP Server: reg. sip-trunk. telekom. de
SIP Server Port: 5060
Context: from-pstn-toheader
Transport: 0.0.0.0-tcp

Advanced

DTMF Mode: RFC 4733
Permanent Auth Rejection: Yes
Forbidden Retry Interval: 30 (Seconds)
Fatal Retry Interval: 0 (Seconds)
General Retry Interval: 60 (Seconds)
Expiration: 300 (Seconds)
Max Retries: 10
Qualify Frequency: 60 (Seconds)
Outbound Proxy: sip:reg. sip-trunk. telekom. de
Contact user: +49zzzzzzzz
From Domain: sip-trunk. telekom. de
From User: (empty)
Client-URI: sip:+49zzzzzzzz @ sip-trunk. telekom. de:5060
Server URI: sip:xxxxxxxxxxxx @ reg.sip-trunk. telekom. de:5060
Media Address: (empty)
AOR: (empty)
AOR Contact: (empty)
Match (Permit): 217.0.0.0/13
Support Path: No
Support T.38 UDPTL: No
T.38 UDPTL Error Correction: None
T.38 UDPTL NAT: No
T.38 UDPTL MAXDATAGRAM: (empty)
Fax Detect: No
Trust RPID/PAI: Yes
Send RPID/PAI: Send P-Asserted-Identity header
Match Inbound Authentication: Standard
Inband Progress: No
Direct Media: No
Rewrite contact: Yes
RTP Symmetric: Yes
Media Encryption: Kein*e
Message Context: (empty)

Codecs

G722
G726
alaw
ulaw
gsm

The phone number +49zzzzzzzz is ONLY the part of your number which is identical for all extensions. No extension number.

You have to delete the spaces in the links. These were necessary because I’m not allowed to post links.

Maybe this helps.


(Michael Fuchs) #5

Hello Trunk,
thank you very much, I’ll give it a try :wink:
Cheers Michael


(Michael Fuchs) #6

Hello Trunk,
I tested it and YEEEEAAAAHHHHH you saved my day, thank you very, very much, this was really troubling me…
Cheers Michael