No incomming calls with IPTAM

Hello everyone,

I’ve got a strange problem with my IPTAM PBX with underlying Asterisk. Incoming calls get rejected, but if you make an outbound call it works for a few minutes.

[general]
udpbindaddr = 0.0.0.0:25060
allowguest = yes
transport = udp
;disallow = all
allow = alaw
allow = ulaw
language = de
useragent = IPTAM PBX (Version 4.0.4/4185)
alwaysauthreject = yes
callcounter = yes
fromdomain = *externalIP*
notifycid = ignore-context
notifyprivacy = no
match_auth_username = yes
directmedia = no
t38pt_udptl = no
t1min = 500
timert1 = 500
nat = force_rport,comedia
tos_sip = 96
tos_audio = 184
allowguest = no
session-timers = originate
externip = "externalIP"
externtcpport =  25060
externtlsport = 25061
localnet=192.168.178.0/24
defaultexpiry = 3600
register => number:pass@host:5060/number

[sip-acc-4]
type=peer
username=number
secret=rfGAeWUs
host=host
port=5060
fromuser=number
fromdomain=host
context=incoming
insecure=very
directmedia=no
canreinvite=yes
allow=ulaw
dtmfmode=inband
allow = alaw
t38pt_udptl = no
setvar=UseFAXt38gw=0

[DTAG-IP_IN16_026](sip-acc-3)
host=IP of host for load balancing reasons
trustrpid=no

Here is a call trace of an incoming call:

<--- SIP read from UDP:ProviderIP8:5060 --->
Cirpack KeepAlive Packet
<------------->

<--- SIP read from UDP:Provider:5060 --->
INVITE sip:Number@ExternalIP:25060 SIP/2.0
Call-ID: x24dABux3:Fu:1XT
Contact: <sip:Provider:5060>
Content-Type: application/sdp
CSeq: 18755 INVITE
From: <sip:NumberIncoming@ims.telekom.de;user=phone>;tag=87.136.5.65+4+7c1d65d4+50c39e7f+BIABNGQ
Max-Forwards: 56
Record-Route: <sip:Provider:5060;transport=udp;lr>;session=18157
Supported: histinfo,replaces,sec-agree,timer,199,100rel
To: <sip:+Number@sipreg3.voice.ewetel.de;user=phone>
Via: SIP/2.0/UDP Provider:5060;branch=z9hG4bK-JUSR-396a32eb-483e88d7
Accept: application/sdp,application/3gpp-ims+xml
Expires: 180
Min-SE: 900
P-Early-Media: supported
Session-Expires: 1920
Session-ID: 1323182645785af6d6de2afbc46108f3;remote=00000000000000000000000000000000
User-Agent: 
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL,INFO,PRACK,NOTIFY,REFER,UPDATE,MESSAGE
History-Info: <sip:+Number@ims.telekom.de;user=phone>;index=1
P-Asserted-Identity: <sip:NumberIncoming6;cpc=mobile-hplmn@Provider:5060;user=phone>, <tel:NumberIncoming6;cpc=mobile-hplmn>
Content-Length: 354

v=0
o=anonymous 110337760607868 110337760607868 IN IP4 Provider
s=-
c=IN IP4 213.168.198.42
t=0 0
m=audio 31498 RTP/AVP 9 8 0 111 110
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 telephone-event/16000
a=fmtp:111 0-15
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
a=ptime:20
a=maxptime:40
a=sendrecv
<------------->
--- (22 headers 16 lines) ---
Sending to Provider:5060 (NAT)
Sending to Provider:5060 (NAT)
Using INVITE request as basis request - x24dABux3:Fu:1XT
Found peer 'DTAG-IP_IN16_026' for 'NumberIncoming' from Provider:5060

<--- Reliably Transmitting (NAT) to Provider:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP Provider:5060;branch=z9hG4bK-JUSR-396a32eb-483e88d7;received=Provider;rport=5060
From: <sip:NumberIncoming@ims.telekom.de;user=phone>;tag=87.136.5.65+4+7c1d65d4+50c39e7f+BIABNGQ
To: <sip:+Number@sipreg3.voice.ewetel.de;user=phone>;tag=as2f3c4916
Call-ID: x24dABux3:Fu:1XT
CSeq: 18755 INVITE
Server: IPTAM PBX (Version 4.0.4/4185)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27167c85"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'x24dABux3:Fu:1XT' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:Provider:5060 --->
ACK sip:Number@ExternalIP:25060 SIP/2.0
Call-ID: x24dABux3:Fu:1XT
CSeq: 18755 ACK
From: <sip:NumberIncoming@ims.telekom.de;user=phone>;tag=87.136.5.65+4+7c1d65d4+50c39e7f+BIABNGQ
Max-Forwards: 56
To: <sip:+Number@sipreg3.voice.ewetel.de;user=phone>;tag=as2f3c4916
Via: SIP/2.0/UDP Provider:5060;branch=z9hG4bK-JUSR-396a32eb-483e88d7
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

Thx for your help

Please convert to chan_pjsip and only specify an outbound authentication.

With chan_sip, you should be using remotesecret, not secret. The “very” option for “insecure” hasn’t been recognized for a very long time and remotesecret is a much cleaner way of doing what insecure=invite is commonly used for. Most people do not need to make the port insecure.

Also note: username doesn’t do what you think it does; canreinvite is an obsolete name for directmedia, so, as it is different, cancel out the directmedia setting; you should always start the codecs with disallow all; and inband DTMF is unusual, but may work here.

as the IPTAM PBX is a commercial product, you should first contact your supplier’s support.

Manual changes are overwritten by automatic processes.

For the problem with rejected incoming calls, you should add the word “viele” to the SIP server settings under "Weitere SIP-Server ".

HTH

Best regards

Karsten

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