Good day,
I´m starting to be frustated 
Running version 9.0.1. and have problems configuring Sip-trunks, esp sipgate.de trunks.
Trunks are fine for outgoing calls, no problem so far. And they are registered with the provider.
But incoming does not work at all. Sitting here for approx 6 hours to get it running, but no chance. Tried all configurations I googled so far, still without any result.
Anybody out there who knows how it works?
Cheer
I can see the status of my trunks using sip show peers.
Unless you know of a scenario, where the trunk is unregistered, but still reports status as ‘OK’, in which case I’m all ears.
Cheers
I agree, do show the verbose log to see if anything is hitting.
A point of order “sip show peers” does not show registration status. “sip show registry and sip show registrations” are the commands you are looking for.
hi all
I have the same problem with freepbx2.9, asterisk 1.8.3.3 on centos 5.5x64 in a openvz container
when i dial by pstn to my did number ( trunk 052xxxxxxxxxx), i get a fast hangup.
on the console i can see the following:
I was using the same trunk configuration with a container openvz with centos 5.5x64, trixbox 2.8 and asterisk 1.6 that works fine.
<— SIP read from UDP:83.211.227.21:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: sip:83.211.227.21;lr=on;ftag=A7D97448-8D1
Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK6eb5.5820a67.0
Via: SIP/2.0/UDP 83.211.2.218:5060;rport=61400;received=83.211.2.218;x-route-tag=“tgrp:Slot6”;branch=z9hG4bKD434F16C
From: sip:[email protected];tag=A7D97448-8D1
To: sip:[email protected]
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Max-Forwards: 9
Remote-Party-ID: sip:[email protected];party=calling;screen=no;privacy=off
Contact: sip:[email protected]:61400
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 357
P-hint: 2 Niente 2
v=0
o=CiscoSystemsSIP-GW-UserAgent 30 3075 IN IP4 83.211.2.218
s=SIP Call
c=IN IP4 62.94.199.36
t=0 0
m=audio 50762 RTP/AVP 18 8 0 4 3 125
c=IN IP4 62.94.199.36
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=5.3;annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:125 X-CCD/8000
<------------->
— (17 headers 15 lines) —
Sending to 83.211.227.21:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘0052xxxxxxxxxx’ for ‘52xxxxxxxxxx’ from 83.211.227.21:5060
<— Reliably Transmitting (NAT) to 83.211.227.21:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 83.211.227.21;branch=z9hG4bK6eb5.5820a67.0;received=83.211.227.21;rport=5060
Via: SIP/2.0/UDP 83.211.2.218:5060;rport=61400;received=83.211.2.218;x-route-tag=“tgrp:Slot6”;branch=z9hG4bKD434F16C
From: sip:[email protected];tag=A7D97448-8D1
To: sip:[email protected];tag=as6c700c78
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.9.0(1.8.3.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="17621591"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
Hey, don’t worry about being a beginner, we all started somewhere
As for your trunk, it appears to be registered with sip.de, so I’d suggest hopping in the console, typing asterisk -vr and checking that the trunk is still registered by typing 'sip show peers’
Once you’ve confirmed that the trunk is still registered, stay in the console, and make an inbound call from a cellphone to the DDI of the trunk in question.
If nothing appears in the console, then I would suggest speaking with sip.de, as they may have not paired the DDI to your sip details.
Just out of interest, are you getting any calls through your other sip.de trunks ?
I noticed you have several with them, so I’m assuming you’ve been able to get both inbound/outbound service on them ? Or am I jumping ahead here ?
Do you actually have DDI/DID’s associated with these trunks ?
Good day,
as you might have noticed, I am a bloody beginner with FreePBX, so my settings are just copied from the providers web-site or different forum entries.
I really appriciate your help.
So I added two codecs and changed the insecure setting to insecure=invite. Unfortunatelly it did not help at all.
Trunks are definitely registered with the provider, as I can see the FreePBX-Server in their web-interface and I can use the trunks for outgoing calls.
The trunk we are talking about is 4318648 and as you can see it is registered with the provider:
Sip show registry:
217.10.72.53:5060 N 4313787 120 Request Sent
217.10.79.9:5060 N 4318648 105 Registered Mon, 09 May 2011 14:45:42
217.10.79.23:5060 N 1008242 105 Registered Mon, 09 May 2011 14:45:42
217.10.79.9:5060 N 8706563e0 120 Auth. Sent
sipgate.de:5060 N 5931322 105 Registered Mon, 09 May 2011 14:45:43
83.125.8.83:5060 N 000387827100 105 Registered Mon, 09 May 2011 14:45:43
83.125.8.83:5060 N 000387804608 105 Registered Mon, 09 May 2011 14:45:43
83.125.8.83:5060 N 000387225768 105 Registered Mon, 09 May 2011 14:45:43
83.125.8.83:5060 N 000387805955 105 Registered Mon, 09 May 2011 14:45:43
217.10.79.9:5060 N 1148546 105 Registered Mon, 09 May 2011 14:45:43
217.10.79.9:5060 N 6043868 105 Registered Mon, 09 May 2011 14:45:43
sipgate.de:5060 N 1103813e1 105 Registered Mon, 09 May 2011 14:45:43
82.117.59.132:5060 N 48327976721 120 Request Sent
13 SIP registrations.
You can’t use insecure=very and still have usernames and secrets.
Now you don’t have any CODEC’s !!
Why do you need the username and the from-domain statement?
The context from-sipgate is invalid, should be from-trunk
Are you registered to your provider? You must be to get any traffic.
Please send the output of “sip show registrations” and “sip show registry”
unfortunatelly there is pretty much nothing to see in the asterisk console:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
thats all.
configuration as suggested by sipgate
peer details
fromdomain=sipgate.de
allow=alaw&ulaw&g729&gsm&slinear
canredirect=no
canreinvite=yes
disallow=all
host=217.10.79.9 => this is sipgate.de
insecure=very
secret=
type=friend
username=
authuser=
context=ext-did
fromuser=
USER DETAILS
type=user
secret=
context=from-trunk
qualify=yes
registration string
:@217.10.79.9/
Tried different setting as found in different posts, but none of them is working.
GENERAL SETTINGS
Allow anonymous inbound SIP Calls “yes”
FreePBS is running on a XENTOS virtual server, not sure about the Linux distribution. Installed Version is Asterisk Now 1.7.1, but updated to FreePBX 2.9.0.1.
Problems ONLY with the sipgate trunks, we do have a lot of other SIP-trunks, running without any problems.
Cheers
Karsten
well, you have the disallow after the allow so no CODEC’s will be allowed.
Linear is not a CODEC
Do you have g.729 licenses installed if not can’t use g.729
Is the trunk actually registered (look in asterisk info)
I doubt you need insecure=very
canreinvite should be note except in certain circumstances.
The settings look like a scattered mess. Try those changes to start.
you are right, settings are a little bit confusing. So I streamlined them and changed them to
type=friend
insecure=very
nat=yes
username=SIP-ID
fromuser=SIP-ID
fromdomain=sipgate.de
secret=SIP-PW
host=217.10.79.9 --> sipgate.de
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context=from-sipgate
unfortunatelly it did not help at all.
here again the entries found in the asterisk console, when I tried to make a call. for me it looks like only the extensions registered with the asterisk swerver, but can not see anything else. just one trunk activated, all other trunks deactivated. Call coming from external route.
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
<— SIP read from UDP:91.64.186.3:5060 —>
<------------->
<— SIP read from UDP:60.241.4.3:51400 —>
<------------->
<— SIP read from UDP:217.91.13.70:5061 —>
<------------->
<— SIP read from UDP:60.241.4.3:5060 —>
<------------->
<— SIP read from UDP:83.10.2.193:18501 —>
<------------->
<— SIP read from UDP:91.64.182.250:35992 —>
<------------->
<— SIP read from UDP:91.64.186.3:5060 —>
REGISTER sip:80.190.117.194 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK640b98fd699f540d40c99bae6d34d229;rport
From: “Andreas Spremberg” sip:[email protected];tag=942438766
To: “Andreas Spremberg” sip:[email protected]
Call-ID: 2859221348@192_168_0_101
CSeq: 18727 REGISTER
Contact: sip:[email protected]:5060
Max-Forwards: 70
User-Agent: C470IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Sending to 192.168.0.101 : 5060 (no NAT)
<— Transmitting (NAT) to 91.64.186.3:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK640b98fd699f540d40c99bae6d34d229;received=91.64.186.3;rport=5060
From: “Andreas Spremberg” sip:[email protected];tag=942438766
To: “Andreas Spremberg” sip:[email protected];tag=as3df965d8
Call-ID: 2859221348@192_168_0_101
CSeq: 18727 REGISTER
Server: FPBX-2.9.0(1.6.2.17.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="63913950"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘2859221348@192_168_0_101’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:91.64.186.3:5060 —>
REGISTER sip:80.190.117.194 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK6fe6439d7bbf800dc4dc0285ce53d7ab;rport
From: “Andreas Spremberg” sip:[email protected];tag=942438766
To: “Andreas Spremberg” sip:[email protected]
Call-ID: 2859221348@192_168_0_101
CSeq: 18728 REGISTER
Contact: sip:[email protected]:5060
Authorization: Digest username=“10”, realm=“asterisk”, algorithm=MD5, uri=“sip:80.190.117.194”, nonce=“63913950”, response="f69cb4562b7911724cae34a608989a3f"
Max-Forwards: 70
User-Agent: C470IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 91.64.186.3 : 5060 (NAT)
Reliably Transmitting (NAT) to 91.64.186.3:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 80.190.117.194:5060;branch=z9hG4bK16c61b47;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as31ba3bda
To: sip:[email protected]:5060
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.17.3)
Date: Sun, 08 May 2011 03:13:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<— Transmitting (NAT) to 91.64.186.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK6fe6439d7bbf800dc4dc0285ce53d7ab;received=91.64.186.3;rport=5060
From: “Andreas Spremberg” sip:[email protected];tag=942438766
To: “Andreas Spremberg” sip:[email protected];tag=as3df965d8
Call-ID: 2859221348@192_168_0_101
CSeq: 18728 REGISTER
Server: FPBX-2.9.0(1.6.2.17.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: sip:[email protected]:5060;expires=180
Date: Sun, 08 May 2011 03:13:43 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘2859221348@192_168_0_101’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:91.64.186.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.190.117.194:5060;branch=z9hG4bK16c61b47;rport=5060
From: “Unknown” sip:[email protected];tag=as31ba3bda
To: sip:[email protected]:5060;tag=1955993965
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:[email protected]:5060
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
<— SIP read from UDP:60.241.4.3:51400 —>
REGISTER sip:80.190.117.194 SIP/2.0
Via: SIP/2.0/UDP 60.241.4.3:49164;branch=z9hG4bKedbb9c9a7b52fbfbdbd643c838b668db;rport
From: “Karsten Knorr” sip:[email protected];tag=2013248124
To: “Karsten Knorr” sip:[email protected]
Call-ID: 1823606155@10_0_0_11
CSeq: 2028 REGISTER
Contact: sip:[email protected]:49164
Max-Forwards: 70
User-Agent: S675IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Sending to 60.241.4.3 : 49164 (no NAT)
<— Transmitting (NAT) to 60.241.4.3:51400 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 60.241.4.3:49164;branch=z9hG4bKedbb9c9a7b52fbfbdbd643c838b668db;received=60.241.4.3;rport=51400
From: “Karsten Knorr” sip:[email protected];tag=2013248124
To: “Karsten Knorr” sip:[email protected];tag=as3cb796e0
Call-ID: 1823606155@10_0_0_11
CSeq: 2028 REGISTER
Server: FPBX-2.9.0(1.6.2.17.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="65cffbb8"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘1823606155@10_0_0_11’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:60.241.4.3:51400 —>
REGISTER sip:80.190.117.194 SIP/2.0
Via: SIP/2.0/UDP 60.241.4.3:49164;branch=z9hG4bK2a8fe1404cf48bd4e858eefc9a66595c;rport
From: “Karsten Knorr” sip:[email protected];tag=2013248124
To: “Karsten Knorr” sip:[email protected]
Call-ID: 1823606155@10_0_0_11
CSeq: 2029 REGISTER
Contact: sip:[email protected]:49164
Authorization: Digest username=“20”, realm=“asterisk”, algorithm=MD5, uri=“sip:80.190.117.194”, nonce=“65cffbb8”, response="d59c524e0abc71cf3b8081e0c254800b"
Max-Forwards: 70
User-Agent: S675IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 60.241.4.3 : 51400 (NAT)
Reliably Transmitting (NAT) to 60.241.4.3:51400:
OPTIONS sip:[email protected]:49164 SIP/2.0
Via: SIP/2.0/UDP 80.190.117.194:5060;branch=z9hG4bK1f139e68;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as183bcffc
To: sip:[email protected]:49164
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.17.3)
Date: Sun, 08 May 2011 03:13:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<— Transmitting (NAT) to 60.241.4.3:51400 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 60.241.4.3:49164;branch=z9hG4bK2a8fe1404cf48bd4e858eefc9a66595c;received=60.241.4.3;rport=51400
From: “Karsten Knorr” sip:[email protected];tag=2013248124
To: “Karsten Knorr” sip:[email protected];tag=as3cb796e0
Call-ID: 1823606155@10_0_0_11
CSeq: 2029 REGISTER
Server: FPBX-2.9.0(1.6.2.17.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: sip:[email protected]:49164;expires=180
Date: Sun, 08 May 2011 03:13:45 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘1823606155@10_0_0_11’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:60.241.4.3:51400 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.190.117.194:5060;branch=z9hG4bK1f139e68;rport=5060
From: “Unknown” sip:[email protected];tag=as183bcffc
To: sip:[email protected]:5060;tag=2709996533
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:[email protected]:5060
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
<— SIP read from UDP:60.241.4.3:5060 —>
REGISTER sip:80.190.117.194 SIP/2.0
Via: SIP/2.0/UDP 60.241.4.3:49158;branch=z9hG4bK5b022d64fb95e30f5bbfeeb41c874096;rport
From: “Sabine & Karsten privat” sip:[email protected];tag=3099221857
To: “Sabine & Karsten privat” sip:[email protected]
Call-ID: 270000386@10_0_0_12
CSeq: 2027 REGISTER
Contact: sip:[email protected]:49158
Max-Forwards: 70
User-Agent: A580 IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Sending to 60.241.4.3 : 49158 (no NAT)
<— Transmitting (NAT) to 60.241.4.3:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 60.241.4.3:49158;branch=z9hG4bK5b022d64fb95e30f5bbfeeb41c874096;received=60.241.4.3;rport=5060
From: “Sabine & Karsten privat” sip:[email protected];tag=3099221857
To: “Sabine & Karsten privat” sip:[email protected];tag=as549d224f
Call-ID: 270000386@10_0_0_12
CSeq: 2027 REGISTER
Server: FPBX-2.9.0(1.6.2.17.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5b8d36e8"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘270000386@10_0_0_12’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:60.241.4.3:5060 —>
REGISTER sip:80.190.117.194 SIP/2.0
Via: SIP/2.0/UDP 60.241.4.3:49158;branch=z9hG4bKcc1605ad350cadb06feca2661153a60d;rport
From: “Sabine & Karsten privat” sip:[email protected];tag=3099221857
To: “Sabine & Karsten privat” sip:[email protected]
Call-ID: 270000386@10_0_0_12
CSeq: 2028 REGISTER
Contact: sip:[email protected]:49158
Authorization: Digest username=“11”, realm=“asterisk”, algorithm=MD5, uri=“sip:80.190.117.194”, nonce=“5b8d36e8”, response="07bfe2a6668fc5b794126d35ac51955c"
Max-Forwards: 70
User-Agent: A580 IP/022270000000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 60.241.4.3 : 5060 (NAT)
Reliably Transmitting (NAT) to 60.241.4.3:5060:
OPTIONS sip:[email protected]:49158 SIP/2.0
Via: SIP/2.0/UDP 80.190.117.194:5060;branch=z9hG4bK09847687;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as3851e7be
To: sip:[email protected]:49158
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.17.3)
Date: Sun, 08 May 2011 03:13:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<— Transmitting (NAT) to 60.241.4.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 60.241.4.3:49158;branch=z9hG4bKcc1605ad350cadb06feca2661153a60d;received=60.241.4.3;rport=5060
From: “Sabine & Karsten privat” sip:[email protected];tag=3099221857
To: “Sabine & Karsten privat” sip:[email protected];tag=as549d224f
Call-ID: 270000386@10_0_0_12
CSeq: 2028 REGISTER
Server: FPBX-2.9.0(1.6.2.17.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 180
Contact: sip:[email protected]:49158;expires=180
Date: Sun, 08 May 2011 03:13:47 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘270000386@10_0_0_12’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:60.241.4.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.190.117.194:5060;branch=z9hG4bK09847687;rport=5060
From: “Unknown” sip:[email protected];tag=as3851e7be
To: sip:[email protected]:5060;tag=1212070361
Call-ID: [email protected]
CSeq: 102 OPTIONS
Contact: sip:[email protected]:5060
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
Really destroying SIP dialog ‘462395085@10_0_1_3’ Method: REGISTER
<— SIP read from UDP:217.91.13.70:5061 —>
<------------->
<— SIP read from UDP:83.10.2.193:18501 —>
<------------->
<— SIP read from UDP:91.64.182.250:35992 —>
How could we possibly help you if you did not tell us what you did?
We also don’t have a version 2.9.0.1
Read “how to ask for help”
Need to know what Asterisk you are running, how you installed and the trunk configs you tried (sanitized). Also verbose logs and SIP traces are a big help.
If you post long one use pastebin.ca and link back to them in your post.
Well, to start with the obvious - have you set up inbound routes ?
Also, when you try calling into your PBX, do you see the call hitting your PBX, when you’re in the asterisk -vr console ?
Answering those questions will allow us to identify whether it’s a misconfiguration at the PBX or your router.
I appreciate your frustration with this, but I’d suggest giving as much information as possible when seeking assistance - it’s like going to the doctors, and expecting a diagnosis because you ‘don’t feel well’.
hi
searching i found the following info, and works
The peer section in a trunk need have this info
host=ip.address.of.host.that.is.sending.SIP.invites
type=friend
insecure=port,invite
dissallow=all
allow=ulaw
context=from-trunk
Regards
Victor