[asterisk-users] feeling n00b again

asterisk at a-domani.nl asterisk at a-domani.nl
Mon Aug 27 18:55:15 CDT 2018


Retrying, falling of the list some how :-(


-------- Original Message --------
Subject: feeling n00b again
Date: 2018-08-20 09:51
 From: asterisk at a-domani.nl
To: asterisk-users at lists.digium.com

Hi all,

Long time ago, I followed a Asterisk training, and both at work and at 
home, was able to deploy Asterisk,
make all sorts of internal call (hard/soft voip-phones, 
incoming/outgoing, ISDN-BA/PRA, SIP, IAX2, ata's, realtime, voicemail, 
load-balancing crypto; almost anything.
But the days of the 1.2-release is long gone.

Now, I'm trying to pick it up again, but even the most simple config 
seems to fail.
Hardly anything seems to works.


Situation:
Asterisk 15.2.2 on a x86_64 running Linux on 2018-03-06 15:11:19 UTC
phone1: Grandsteam2000
phone2: Siemens DECT
phone are all on same subnet (no nat involved here)

1) Echo functionality on phone 1: OK (using alaw codec)
2) Echo functionality on phone 2: OK (using alaw codec)
3) Call from  phone2 to phone1: OK (both using alaw)
4) Call from phone1 to phone2: immediate disconnect after answering
(might not be related) console says:

[Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10434 
process_sdp: Received AVP profile in audio answer but AVPF is enabled: 
audio 7200 RTP/AVP 8 101
[Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819 
process_sdp: Failing due to no acceptable offer found

I enabled debug on the IP of the dect-phone (full log attached), but it 
does not make me any wiser...


set_destination: Parsing <sip:dect at 192.168.0.27:5060> for address/port 
to send to
set_destination: set destination to 192.168.0.27:5060
Reliably Transmitting (no NAT) to 192.168.0.27:5060:
BYE sip:dect at 192.168.0.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8
Max-Forwards: 70
 From: "fam-witvliet eerste verdiep" 
<sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 15.2.2
===> X-Asterisk-HangupCause: Bearer capability not available     <===
===> X-Asterisk-HangupCauseCode: 58                              <===
Content-Length: 0



Anyone around to give some pointers/clues?
-------------- next part --------------

[Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10434 process_sdp: Received AVP profile in audio answer but AVPF is enabled: audio 7200 RTP/AVP 8 101
[Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819 process_sdp: Failing due to no acceptable offer found

voip4-01/voip4-01         192.168.0.33
dect/dect                 192.168.0.27 

pbx*CLI> sip set debug ip 192.168.0.27
SIP Debugging Enabled for IP: 192.168.0.27


pbx*CLI> 
pbx*CLI> 
Audio is at 18886
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.27:5060:
INVITE sip:dect at 192.168.0.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK7572c876
Max-Forwards: 70
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>
Contact: <sip:voip4-01 at 192.168.0.25:5060>
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.2.2
Date: Mon, 20 Aug 2018 07:25:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1497526820 1497526820 IN IP4 192.168.0.25
s=Asterisk PBX 15.2.2
c=IN IP4 192.168.0.25
t=0 0
m=audio 18886 RTP/AVPF 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
Retransmitting #1 (no NAT) to 192.168.0.27:5060:
INVITE sip:dect at 192.168.0.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK7572c876
Max-Forwards: 70
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>
Contact: <sip:voip4-01 at 192.168.0.25:5060>
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.2.2
Date: Mon, 20 Aug 2018 07:25:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1497526820 1497526820 IN IP4 192.168.0.25
s=Asterisk PBX 15.2.2
c=IN IP4 192.168.0.25
t=0 0
m=audio 18886 RTP/AVPF 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.168.0.27:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK7572c876
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 102 INVITE
Contact: <sip:dect at 192.168.0.27:5060>
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.27:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK7572c876
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 102 INVITE
Contact: <sip:dect at 192.168.0.27:5060>
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.27:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK7572c876
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 102 INVITE
Contact: <sip:dect at 192.168.0.27:5060>
Allow-Events: message-summary, refer
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:dect at 192.168.0.27:5060>

<--- SIP read from UDP:192.168.0.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK7572c876
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 102 INVITE
Contact: <sip:dect at 192.168.0.27:5060>
Supported: replaces
Allow-Events: message-summary, refer
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 196

v=0
o=dect 7214 19 IN IP4 192.168.0.27
s=Mapping
c=IN IP4 192.168.0.27
t=0 0
m=audio 7214 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (12 headers 10 lines) ---
[Aug 20 09:25:12] WARNING[7080][C-00000120]: chan_sip.c:10434 process_sdp: Received AVP profile in audio answer but AVPF is enabled: audio 7214 RTP/AVP 8 101
[Aug 20 09:25:12] WARNING[7080][C-00000120]: chan_sip.c:10819 process_sdp: Failing due to no acceptable offer found
sip_route_dump: route/path hop: <sip:dect at 192.168.0.27:5060>
set_destination: Parsing <sip:dect at 192.168.0.27:5060> for address/port to send to
set_destination: set destination to 192.168.0.27:5060
Transmitting (no NAT) to 192.168.0.27:5060:
ACK sip:dect at 192.168.0.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK6c6973bb
Max-Forwards: 70
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Contact: <sip:voip4-01 at 192.168.0.25:5060>
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.2.2
Content-Length: 0


---
set_destination: Parsing <sip:dect at 192.168.0.27:5060> for address/port to send to
set_destination: set destination to 192.168.0.27:5060
Reliably Transmitting (no NAT) to 192.168.0.27:5060:
BYE sip:dect at 192.168.0.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8
Max-Forwards: 70
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 15.2.2
===> X-Asterisk-HangupCause: Bearer capability not available     <===
===> X-Asterisk-HangupCauseCode: 58                              <===
Content-Length: 0


---
Scheduling destruction of SIP dialog '78d92db820b4926879361f7d4968444a at 192.168.0.25:5060' in 7040 ms (Method: INVITE)
Scheduling destruction of SIP dialog '78d92db820b4926879361f7d4968444a at 192.168.0.25:5060' in 7040 ms (Method: INVITE)
Retransmitting #1 (no NAT) to 192.168.0.27:5060:
BYE sip:dect at 192.168.0.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8
Max-Forwards: 70
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 15.2.2
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 103 BYE
Contact: <sip:dect at 192.168.0.27:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '78d92db820b4926879361f7d4968444a at 192.168.0.25:5060' Method: INVITE

<--- SIP read from UDP:192.168.0.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8
From: "fam-witvliet eerste verdiep" <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
CSeq: 103 BYE
Contact: <sip:dect at 192.168.0.27:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '625250224 at 192_168_0_27' Method: REGISTER
pbx*CLI> 
pbx*CLI> sip set debug off
SIP Debugging Disabled
pbx*CLI> 




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