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asterisk at a-domani.nl

2018-Aug-27 23:55 UTC

[asterisk-users] feeling n00b again

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>

John Kiniston

2018-Aug-28 16:28 UTC

head link

[asterisk-users] feeling n00b again

It's a configuration issue with the peer (or the peer it's matching) in
sip.conf

You have configured the peer to use avpf but your phone is not attempting
to do avpf.

Either configure the phone to match the entry in sip.conf or change
sip.conf to match the phone.


On Mon, Aug 27, 2018 at 4:56 PM <asterisk at a-domani.nl> wrote:
> 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?--
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
-------------- next part --------------
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asterisk at a-domani.nl

2018-Aug-29 16:03 UTC

head link

[asterisk-users] feeling n00b again

Thanks John

I'll try tonight with avpf=no, and see what will happen

Hans

On 2018-08-28 18:28, John Kiniston wrote:> It's a configuration issue with the peer (or the peer it's
matching)
> in sip.conf
> 
> You have configured the peer to use avpf but your phone is not
> attempting to do avpf.
> 
> Either configure the phone to match the entry in sip.conf or change
> sip.conf to match the phone.
> 
> On Mon, Aug 27, 2018 at 4:56 PM <asterisk at a-domani.nl> wrote:
> 
>> 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 [1]> for
>> address/port
>> to send to
>> set_destination: set destination to 192.168.0.27:5060 [2]
>> Reliably Transmitting (no NAT) to 192.168.0.27:5060 [2]:
>> BYE sip:dect at 192.168.0.27:5060 [1] 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 [1]>;tag=1813732733
>> Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060 [3]
>> 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?--
>> 
> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
>> --
>> 
>> Astricon is coming up October 9-11!  Signup is available at:
>> https://www.asterisk.org/community/astricon-user-conference
>> 
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>> 
>> New to Asterisk? Start here:
>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> A human being should be able to change a diaper, plan an invasion,
> butcher a hog, conn a ship, design a building, write a sonnet, balance
> accounts, build a wall, set a bone, comfort the dying, take orders,
> give orders, cooperate, act alone, solve equations, analyze a new
> problem, pitch manure, program a computer, cook a tasty meal, fight
> efficiently, die gallantly. Specialization is for insects.
> ---Heinlein
> 
> Links:
> ------
> [1] http://sip:dect at 192.168.0.27:5060
> [2] http://192.168.0.27:5060
> [3] http://78d92db820b4926879361f7d4968444a at 192.168.0.25:5060

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