Connecting two softphones to cameras without success!

I installed Asterisk 16.1.1 built by root @xxx on a x86_64 running Linux on 2019-02-11 18:45:31 UTC

I setting up the sip.conf, pjsip.conf and extensions.conf files. I want to connect two Zoiper softphones and 2 cameras grandstream.

But I cannot to call between the softphones to the cameras, neither softphones between them!

I follow all recommendation of setting, but I cannot to do it works!

Could you help me with this?

The errors when I open asterisk from command line: CLI are:

[Feb 16 12:19:20] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘sip:6004@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:1025’ (callid: 1119441443-5060-2@BJC.BGI.II.BGH) - Failed to authenticate
[Feb 16 12:19:20] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘sip:6004@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:1025’ (callid: 1119441443-5060-2@BJC.BGI.II.BGH) - No matching endpoint found
[Feb 16 12:19:20] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘sip:6004@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:1025’ (callid: 1119441443-5060-2@BJC.BGI.II.BGH) - Failed to authenticate
[Feb 16 12:19:20] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘sip:6002@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:5060’ (callid: 1859694680-5060-2@BJC.BGI.II.CBA) - No matching endpoint found
[Feb 16 12:19:20] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘sip:6002@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:5060’ (callid: 1859694680-5060-2@BJC.BGI.II.CBA) - Failed to authenticate
[Feb 16 12:19:20] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘sip:6002@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:5060’ (callid: 1859694680-5060-2@BJC.BGI.II.CBA) - No matching endpoint found
[Feb 16 12:19:20] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘sip:6002@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:5060’ (callid: 1859694680-5060-2@BJC.BGI.II.CBA) - Failed to authenticate
[Feb 16 12:19:27] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘“6003” sip:6003@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:10485’ (callid: yQGOz9rp7EKEw1dzou0xZQ…) - No matching endpoint found
[Feb 16 12:19:27] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘“6003” sip:6003@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:10485’ (callid: yQGOz9rp7EKEw1dzou0xZQ…) - No matching endpoint found
[Feb 16 12:19:27] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘“6003” sip:6003@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:10485’ (callid: yQGOz9rp7EKEw1dzou0xZQ…) - Failed to authenticate
[Feb 16 12:19:29] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘“6001” sip:6001@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:25851’ (callid: B62c2XrSYXrw7SDL5YUpZw…) - No matching endpoint found
[Feb 16 12:19:29] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘“6001” sip:6001@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:25851’ (callid: B62c2XrSYXrw7SDL5YUpZw…) - No matching endpoint found
[Feb 16 12:19:29] NOTICE[10496]: res_pjsip/pjsip_distributor.c:672 log_failed_request: Request ‘REGISTER’ from ‘“6001” sip:6001@xxxxxxxxxx’ failed for ‘xxxxxxxxxx:25851’ (callid: B62c2XrSYXrw7SDL5YUpZw…) - Failed to authenticate

The files settings are:

sip.conf

[general]
context=public
disallow=all
allow = alaw,ulaw,g711,g723,h263,gsm,g729
limitonpeers=yes
rtptimeout=600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas
rtptimeout=600
notifyringing=notinuse
notifyhold=yes
callcounter=yes
allowsubscribe=yes
port=5060
bindaddr=0.0.0.0
language=es
[6001]
type=friend
context=from-internal
host=dynamic
secret=***************
[6002]
type=friend
context=from-internal
host=dynamic
secret=***************
[6003]
type=friend
context=from-internal
host=dynamic
secret=***************
[6004]
type=friend
context=from-internal
host=dynamic
secret=***************

extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=yes
[from-internal]
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()
exten=>6001,1,Dial(SIP/6001,30)
exten=>6001,2,Hangup
exten=>6003,1,Dial(SIP/6003,30)
exten=>6003,2,Hangup
exten=>6004,1,Dial(SIP/6004,30)
exten=>6004,2,Hangup

pjsip.conf

[transport-udp]
type=transport
bind=0.0.0.0

[6001]
type=auth
auth_type=userpass
password=***************
username=6001
transport=transport-udp
max_contacts=1
contact=sip:6001@xx.xx.xx.xx:5060
context=from-internal
auth=6001
aors=6001

[6002]
type=auth
auth_type=userpass
password=***************
username=6002
transport=transport-udp
max_contacts=1
contact=sip:6002@xx.xx.xx.xx:5060
context=from-internal
auth=6002
aors=6002

[6003]
type=auth
auth_type=userpass
password=***************
username=6003
transport=transport-udp
max_contacts=1
contact=sip:6002@xx.xx.xx.xx:5060
context=from-internal
auth=6003
aors=6003

[6004]
type=auth
auth_type=userpass
password=***************
username=6004
transport=transport-udp
max_contacts=1
contact=sip:6004@xx.xx.xx.xx:5060
context=from-internal
auth=6004
aors=6004

The phones have been configured with the wrong credentials.

With wrong credentials, you means to bad password?
I have reviewed several times, the passwords are the same in settings files, and the SoftPhones and cameras.
What exactly is “wrong credentials”?

The information used to authenticate is not that which the other side expects.

However, looking more closely, it would appear that the initial problem was that the peer was not recognized as a known peer, as the matching endpoint message comes first. Obviously you cannot authenticate if you can’t work out what is trying to authenticate.

Also, why did you provide sip.conf. It is obviously not being used, and enabling both it and pjsip.conf doesn’t normally work, as only one can bind to port 5060.

I’m not a pjsip user, but you don’t seem to have any endpoints or matches configures at all.

I changed somethings things in sips.conf ,extensions,conf and pjsip.conf.
With good progress but still with failures :frowning:

sip.conf

[general]
context=public
disallow=all
allow = alaw,ulaw,g711,g723,h263,gsm,g729
limitonpeers=yes
rtptimeout=600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas
rtptimeout=600
notifyringing=notinuse
notifyhold=yes
callcounter=yes
allowsubscribe=yes
port=5060
bindaddr=0.0.0.0
language=es
[6001]
type=friend
context=from-internal
host=dynamic
secret=r***E ; Only showing part of password
[6002]
type=friend
context=from-internal
host=dynamic
secret=r***E;Only showing part of password
[6003]
type=friend
context=from-internal
host=dynamic
secret=r***E;Only showing part of password
[6004]
type=friend
context=from-internal
host=dynamic
secret=r***E;Only showing part of password


extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=yes
[from-internal]
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()
exten=>6001,1,Dial(SIP/6001,30)
exten=>6001,2,Hangup
exten=>6002,1,Dial(SIP/6002,30)
exten = 6002,n,Answer()
exten=>6002,n,Hangup
exten=>6003,1,Dial(SIP/6003,30)
exten=>6003,2,Hangup
exten=>6004,1,Dial(SIP/6004,30)
exten = 6004,n,Answer()
exten=>6004,n,Hangup


pjsip.conf

;[transport-udp]
;type=transport
;bind=0.0.0.0
[6001]
type=auth
auth_type=userpass
password=r***E;Only showing part of password
username=6001
transport=udp,tcp
max_contacts=1
contact=sip:6001@xx.xx.xx.xx:5060
context=from-internal
auth=6001
aors=6001
allow=alow,ulow,g711
[6002]
type=auth
auth_type=userpass
password=r***E;Only showing part of password
username=6002
transport=udp,tcp
max_contacts=1
contact=sip:6002@xx.xx.xx.xx:5060
context=from-internal
auth=6002
aors=6002
allow=alow,ulow,g711
[6003]
type=auth
auth_type=userpass
password=r***E;Only showing part of password
username=6003
transport=udp,tcp
max_contacts=1
contact=sip:6003@xx.xx.xx.xx:5060
context=from-internal
auth=6003
aors=6003
allow=alow,ulow,g711
[6004]
type=auth
auth_type=userpass
password=r***E;Only showing part of password
username=6004
transport=udp,tcp
max_contacts=1
contact=sip:6004@xx.xx.xx.xx:5060
context=from-internal
auth=6004
aors=6004
allow=alow,ulow,g711

The logs in *CLI> are:

Saved useragent “Zoiper rv2.9.2” for peer 6003
Saved useragent “Grandstream GXV3611IR_HD 1.0.3.13” for peer 6002
Saved useragent “Zoiper rv2.9.2” for peer 6001
*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
6001/6001 (Unspecified) D Auto (Yes) No 0 Unmonitored
6002/6002 xx.xx.xx.xx D Auto (Yes) No 5060 Unmonitored
6003/6003 xx.xx.xx.xxx D Auto (Yes) No 10485 Unmonitored
6004/6004 xx.xx.xx.xxx D Auto (No) No 5060 Unmonitored
4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 1 offline]
Note: 6004 is camera as well but does not see any record
You are right; pjsip does not well setting, because found this in the *CLI:


pjsip show settings

Global Settings:

ParameterName : ParameterValue

contact_expiration_check_interval : 30
debug : no
default_from_user : asterisk
default_outbound_endpoint : default_outbound_endpoint
default_realm : asterisk
default_voicemail_extension :
disable_multi_domain : false
endpoint_identifier_order : ip,username,anonymous
ignore_uri_user_options : false
keep_alive_interval : 90
max_forwards : 70
max_initial_qualify_time : 0
mwi_disable_initial_unsolicited : false
mwi_tps_queue_high : 500
mwi_tps_queue_low : -1
regcontext :
unidentified_request_count : 5
unidentified_request_period : 5
unidentified_request_prune_interval : 30
use_callerid_contact : no
user_agent : Asterisk PBX 16.1.1

System Settings:

ParameterName : ParameterValue

accept_multiple_sdp_answers : false
compact_headers : false
disable_tcp_switch : true
follow_early_media_fork : true
threadpool_auto_increment : 5
threadpool_idle_timeout : 60
threadpool_initial_size : 0
threadpool_max_size : 50
timer_b : 32000
timer_t1 : 500

pjsip list endpoints

No objects found.

Now I can to do calls beetween Zoipers soft phones but when I answer it, I cannot ear nothing.
And when I call from Zoipers softphone to the camera (6002); I setting up the camera auto-answer previously , and in dialplan as well, but time out and nothing I ear.

Logs in *CLI>

-- Channel SIP/6001-00000004 left 'native_rtp' basic-bridge <f327d16b-13e5-4d21-85b1-5347c6f95a45>

== Spawn extension (from-internal, 6002, 1) exited non-zero on ‘SIP/6001-00000004’
– Channel SIP/6002-00000005 left ‘native_rtp’ basic-bridge
> 0x7feac80069c0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
== Using SIP RTP CoS mark 5
> 0x7feb580086a0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:25906
– Executing [6004@from-internal:1] Dial(“SIP/6001-00000006”, “SIP/6004,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6004
– SIP/6004-00000007 is ringing
> 0x7feac80068a0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
– SIP/6004-00000007 answered SIP/6001-00000006
– Channel SIP/6004-00000007 joined ‘simple_bridge’ basic-bridge <1fa7e1c7-94c4-4396-8e1b-6cf670421414>
– Channel SIP/6001-00000006 joined ‘simple_bridge’ basic-bridge <1fa7e1c7-94c4-4396-8e1b-6cf670421414>
> Bridge 1fa7e1c7-94c4-4396-8e1b-6cf670421414: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/6001-00000006’ and ‘SIP/6004-00000007’ - media will flow directly between them
> 0x7feac80068a0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
> 0x7feb580086a0 – Strict RTP switching to RTP target address xx.xx.xx.xx:25906 as source
– Channel SIP/6001-00000006 left ‘native_rtp’ basic-bridge <1fa7e1c7-94c4-4396-8e1b-6cf670421414>
== Spawn extension (from-internal, 6004, 1) exited non-zero on ‘SIP/6001-00000006’
– Channel SIP/6004-00000007 left ‘native_rtp’ basic-bridge <1fa7e1c7-94c4-4396-8e1b-6cf670421414>
> 0x7feac80068a0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
== Using SIP RTP CoS mark 5
> 0x7feb58008420 – Strict RTP learning after remote address set to: xx.xx.xx.xx:38056
– Executing [6002@from-internal:1] Dial(“SIP/6003-00000008”, “SIP/6002,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6002
– SIP/6002-00000009 is ringing
> 0x7feac80068a0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
– SIP/6002-00000009 answered SIP/6003-00000008
– Channel SIP/6002-00000009 joined ‘simple_bridge’ basic-bridge <32824d76-041e-4bbe-a775-d15cc58afd3f>
– Channel SIP/6003-00000008 joined ‘simple_bridge’ basic-bridge <32824d76-041e-4bbe-a775-d15cc58afd3f>
> Bridge 32824d76-041e-4bbe-a775-d15cc58afd3f: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/6003-00000008’ and ‘SIP/6002-00000009’ - media will flow directly between them
> 0x7feac80068a0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
> 0x7feb58008420 – Strict RTP learning after remote address set to: xx.xx.xx.xx:38056
– Channel SIP/6003-00000008 left ‘native_rtp’ basic-bridge <32824d76-041e-4bbe-a775-d15cc58afd3f>
– Channel SIP/6002-00000009 left ‘native_rtp’ basic-bridge <32824d76-041e-4bbe-a775-d15cc58afd3f>
== Spawn extension (from-internal, 6002, 1) exited non-zero on ‘SIP/6003-00000008’
> 0x7feac80068a0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
== Using SIP RTP CoS mark 5
> 0x7feb58017be0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:38056
– Executing [6004@from-internal:1] Dial(“SIP/:6003-0000000a”, “SIP/6004,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6004
– SIP/6004-0000000b is ringing
> 0x7feacc006790 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
– SIP/6004-0000000b answered SIP/6003-0000000a
– Channel SIP/6004-0000000b joined ‘simple_bridge’ basic-bridge
– Channel SIP/6003-0000000a joined ‘simple_bridge’ basic-bridge
> Bridge f7cbea19-6fe4-4204-9809-fd52926d1bbe: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/6003-0000000a’ and ‘SIP/6004-0000000b’ - media will flow directly between them
> 0x7feacc006790 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
> 0x7feb58017be0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:38056
– Channel SIP/6003-0000000a left ‘native_rtp’ basic-bridge
– Channel SIP/6004-0000000b left ‘native_rtp’ basic-bridge
== Spawn extension (from-internal, 6004, 1) exited non-zero on ‘SIP/6003-0000000a’
> 0x7feacc006790 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004

Please delete either sip.conf or pjsip.conf. Although it is possible to have configurations with both, that would be a rather exceptional usage, and simply causes confusion here.

The fact that you have both suggests you do not have a clear understanding of how Asterisk works.

Or use PJSIP or used chan_sip, but why use both, it will cause more confusion to you as still dont have the basic aAsterisk knowledge, Also I dont see any NAT setting on your configuration in order to handle any Natted enviroment

I followed your advice and I deleted one of either sip.conf or pjsip.conf. Assuring me that I do not have both configurations at same time.
I have chosen sip_chan, and I deleted pjsip.conf, the red pill :wink:
And retouch the setting of sip.conf file this is the result:

sip.conf
[general]
context=public
disallow=all
allow = alaw,ulaw,g723,h263,gsm,g729
limitonpeers=yes
rtptimeout=600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas
rtptimeout=600
notifyringing=notinuse
notifyhold=yes
callcounter=yes
allowsubscribe=yes
port=5060
bindaddr=0.0.0.0
language=es
[6001]
type=friend
context=from-internal
host=dynamic
secret=rU***E; no showing password
[6002]
type=friend
context=from-internal
host=dynamic
secret=rU***E; no showing password
[6003]
type=friend
context=from-internal
host=dynamic
secret=rU***E; no showing password
[6004]
type=friend
context=from-internal
host=dynamic
secret=rU***E; no showing password

And the extensions.conf is:

[general]
static=yes
writeprotect=no
clearglobalvars=yes
[from-internal]
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()
exten=>6001,1,Dial(SIP/6001,30)
exten=>6001,2,Hangup
exten=>6002,1,Dial(SIP/6002,30)
exten = 6002,n,Answer()
exten=>6002,n,Hangup
exten=>6003,1,Dial(SIP/6003,30)
exten=>6003,2,Hangup
exten=>6004,1,Dial(SIP/6004,30)
exten = 6004,n,Answer()
exten=>6004,n,Hangup

But the log in Cli asterisk when I do a call between a softphone and the camera are:
*CLI> – Registered SIP ‘6002’ at xx.xx.xx.xx:1025
> Saved useragent “Grandstream GXV3611IR_HD 1.0.3.13” for peer 6002
> Saved useragent “Zoiper rv2.9.2” for peer 6001

No such command ‘s’ (type ‘core show help s’ for other possible commands)
*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
6001/6001 xx.xx.xx.xx D Auto (Yes) No 25851 Unmonitored 6002/6002 xx.xx.xx.xx D Auto (Yes) No 1025 Unmonitored 6003/6003 xx.xx.xx.xx D Auto (Yes) No 9845 Unmonitored 6004/6004 xx.xx.xx.xx D Auto (Yes) No 5060 Unmonitored
4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline]
*CLI> == Using SIP RTP CoS mark 5
> 0x7fa7680082d0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:25906
– Executing [6002@from-internal:1] Dial(“SIP/6001-00000000”, “SIP/6002,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6002
– SIP/6002-00000001 is ringing
> 0x7fa6e40063c0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
– SIP/6002-00000001 answered SIP/6001-00000000
– Channel SIP/6002-00000001 joined ‘simple_bridge’ basic-bridge
– Channel SIP/6001-00000000 joined ‘simple_bridge’ basic-bridge
> Bridge ca06beb5-c5fb-4d4c-bf84-df7d49c5d007: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/6001-00000000’ and ‘SIP/6002-00000001’ - media will flow directly between them
> 0x7fa6e40063c0 – Strict RTP qualifying stream type: audio
> 0x7fa7680082d0 – Strict RTP switching to RTP target address xx.xx.xx.xx:25906 as source
> 0x7fa6e40063c0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
– Channel SIP/6001-00000000 left ‘native_rtp’ basic-bridge
– Channel SIP/6002-00000001 left ‘native_rtp’ basic-bridge
== Spawn extension (from-internal, 6002, 1) exited non-zero on ‘SIP/6001-00000000’
> 0x7fa6e40063c0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:5004
== Using SIP RTP CoS mark 5
> 0x7fa768016bf0 – Strict RTP learning after remote address set to: xx.xx.xx.xx:25906
– Executing [6004@from-internal:1] Dial(“SIP/6001-00000002”, “SIP/6004,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6004
– SIP/6004-00000003 is ringing
> 0x7fa6e4002b00 – Strict RTP learning after remote address set to:xx.xx.xx.xx:5004
– SIP/6004-00000003 answered SIP/6001-00000002
– Channel SIP/6004-00000003 joined ‘simple_bridge’ basic-bridge <8a383678-59c2-4415-8b78-634cadd32b7d>
– Channel SIP/6001-00000002 joined ‘simple_bridge’ basic-bridge <8a383678-59c2-4415-8b78-634cadd32b7d>
> Bridge 8a383678-59c2-4415-8b78-634cadd32b7d: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/6001-00000002’ and ‘SIP/6004-00000003’ - media will flow directly between them
> 0x7fa768016bf0 – Strict RTP switching to RTP target address xx.xx.xx.xx:25906 as source
> 0x7fa6e4002b00 – Strict RTP learning after remote address set to:xx.xx.xx.xx:5004
– Channel SIP/6001-00000002 left ‘native_rtp’ basic-bridge <8a383678-59c2-4415-8b78-634cadd32b7d>
– Channel SIP/6004-00000003 left ‘native_rtp’ basic-bridge <8a383678-59c2-4415-8b78-634cadd32b7d>
== Spawn extension (from-internal, 6004, 1) exited non-zero on ‘SIP/6001-00000002’
> 0x7fa6e4002b00 – Strict RTP learning after remote address set to:xx.xx.xx.xx:5004
== Using SIP RTP CoS mark 5
> 0x7fa768008210 – Strict RTP learning after remote address set to: xx.xx.xx.xx:25906
– Executing [6003@from-internal:1] Dial(“SIP/6001-00000004”, “SIP/6003,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6003
– SIP/6003-00000005 is ringing
> 0x7fa6e400bf40 – Strict RTP learning after remote address set to: 10.200.135.108:38056
– SIP/6003-00000005 answered SIP/6001-00000004
– Channel SIP/6003-00000005 joined ‘simple_bridge’ basic-bridge <93a6680f-1d86-4bc6-bbef-279ad4b11cd5>
– Channel SIP/6001-00000004 joined ‘simple_bridge’ basic-bridge <93a6680f-1d86-4bc6-bbef-279ad4b11cd5>
> Bridge 93a6680f-1d86-4bc6-bbef-279ad4b11cd5: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/6001-00000004’ and ‘SIP/6003-00000005’ - media will flow directly between them
> 0x7fa6e400bf40 – Strict RTP learning after remote address set to: 10.200.135.108:38056
> 0x7fa768008210 – Strict RTP switching to RTP target address xx.xx.xx.xx:25906 as source
– Channel SIP/6001-00000004 left ‘native_rtp’ basic-bridge <93a6680f-1d86-4bc6-bbef-279ad4b11cd5>
– Channel SIP/6003-00000005 left ‘native_rtp’ basic-bridge <93a6680f-1d86-4bc6-bbef-279ad4b11cd5>
== Spawn extension (from-internal, 6003, 1) exited non-zero on ‘SIP/6001-00000004’
[Feb 17 12:47:14] WARNING[12598][C-00000003]: chan_sip.c:24309 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog ‘5193556464cbda115a91a8a67dcbb757@xx.xx.xx.xx:5060’. Giving up.

The problem is still without solve. I know I don’t have a clear understanding of how asterisk works, but I want a basic thing, connect two softphones to communicate with two cameras. For do that I need be a super expert in Asterisk? Please need your help to solve this soon. Thank you in advance.

Either the Zoiper has already cleared the call, or something is modifying the signalling badly.

You need to get a SIP protocol trace of the complete session. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Also, please mark up any logs and configurations as pre-formatted text (using </>).

Hi,

I had uploaded the Debug file following the instuctions of https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information, I hope it is enought.

Sip File configuration:

[general]
context=public
disallow=all
allow = alaw,ulaw,g723,h263,gsm,g729
limitonpeers=yes
rtptimeout=600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas
rtptimeout=600
notifyringing=notinuse
notifyhold=yes
callcounter=yes
allowsubscribe=yes
port=5060
bindaddr=0.0.0.0
language=es
[6001]
type=friend
context=from-internal
host=dynamic
secret=r*****TE
[6002]
type=friend
context=from-internal
host=dynamic
secret=r*****TE
[6003]
type=friend
context=from-internal
host=dynamic
secret=r*****TE
[6004]
type=friend
context=from-internal
host=dynamic
secret=r*****TE

Extension.conf

[general]
static=yes
writeprotect=no
clearglobalvars=yes
[from-internal]
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()
exten=>6001,1,Dial(SIP/6001,30)
exten=>6001,2,Hangup
exten=>6002,1,Dial(SIP/6002,30)
exten = 6002,n,Answer()
exten=>6002,n,Hangup
exten=>6003,1,Dial(SIP/6003,30)
exten=>6003,2,Hangup
exten=>6004,1,Dial(SIP/6004,30)
exten = 6004,n,Answer()
exten=>6004,n,Hangup

If you need anything else, please tell me.

Regardsdebug01_BillyVB2004.txt (384.0 KB)

You need to set “videosupport” to “yes” in the general section. You should also set “directmedia” to “no”.

Thank for your instant reply :slight_smile:

Yes!! I cant to communnicate from softphone to camera, but beetween two softphone I hear only one!.

I added to Sip file

[general]
videosupport=yes
directmedia=no
context=public
disallow=all
allow = alaw,ulaw,g723,h263,gsm,g729
limitonpeers=yes
rtptimeout=600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas
rtptimeout=600
notifyringing=notinuse
notifyhold=yes
callcounter=yes
allowsubscribe=yes
port=5060
bindaddr=0.0.0.0
language=es
[6001]
type=friend
context=from-internal
host=dynamic
secret=r****E
[6002]
type=friend
context=from-internal
host=dynamic
secret=r****E
[6003]
type=friend
context=from-internal
host=dynamic
secret=r****E
[6004]
type=friend
context=from-internal
host=dynamic
secret=r****E

The debug in *CLI> are:

Asterisk Ready.
*CLI> [Feb 18 08:21:20] WARNING[18402]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission PHVPYkyjuDCRLVR0NXQCtA.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Feb 18 08:21:20] WARNING[18402]: chan_sip.c:4143 retrans_pkt: Hanging up call PHVPYkyjuDCRLVR0NXQCtA.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Feb 18 08:21:42] WARNING[18402]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission XIUkbhJtp0miUxjMw0h1cA.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Feb 18 08:21:42] WARNING[18402]: chan_sip.c:4143 retrans_pkt: Hanging up call XIUkbhJtp0miUxjMw0h1cA.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Feb 18 08:22:27] WARNING[18402]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 4F5PK1TcHD2FQcwHe5jJ_Q.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Feb 18 08:22:27] WARNING[18402]: chan_sip.c:4143 retrans_pkt: Hanging up call 4F5PK1TcHD2FQcwHe5jJ_Q.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Feb 18 08:23:01] WARNING[18402]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission QqVKNbk78HZn8BrWOA6fuQ.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Feb 18 08:23:01] WARNING[18402]: chan_sip.c:4143 retrans_pkt: Hanging up call QqVKNbk78HZn8BrWOA6fuQ.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Feb 18 08:23:34] WARNING[18402]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission KxJVah40iLiIlVbsFqH2Yg.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response

Could you support me with too this please?
Thank in advance!

Is anyone behind NAT? Asterisk? Remote phone?

The messages seem to indicate, at least, that one side can’t talk back to Asterisk.

The Asterisk is in a public IP (with out Firewall) and not is behind none NAT.
The cameras are behind nat, and Zoipers are in a mobile network! (Softphones)
I added a fifth user (a third camera) and now return to before to add the next lines at sip.conf file:

videosupport=yes
directmedia=no

not works anything! :frowning:

this is the sip file now after insert the last user (third camera):

[general]
videosupport=yes
directmedia=no
context=public
disallow=all
allow = alaw,ulaw,g723,h263,gsm,g729
limitonpeers=yes
rtptimeout=600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas
rtptimeout=600
notifyringing=notinuse
notifyhold=yes
callcounter=yes
allowsubscribe=yes
port=5060
bindaddr=0.0.0.0
language=es
[6001]
type=friend
context=from-internal
host=dynamic
secret=r***E
[6002]
type=friend
context=from-internal
host=dynamic
secret=r***E
[6003]
type=friend
context=from-internal
host=dynamic
secret=r***E
[6004]
type=friend
context=from-internal
host=dynamic
secret=r***E
[6005]
type=friend
context=from-internal
host=dynamic
secret=r***E

This is the new extension file after add the last user:

[general]
static=yes
writeprotect=no
clearglobalvars=yes
[from-internal]
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()
exten=>6001,1,Dial(SIP/6001,30)
exten=>6001,2,Hangup
exten=>6002,1,Dial(SIP/6002,30)
exten = 6002,n,Answer()
exten=>6002,n,Hangup
exten=>6003,1,Dial(SIP/6003,30)
exten=>6003,2,Hangup
exten=>6004,1,Dial(SIP/6004,30)
exten = 6004,n,Answer()
exten=>6004,n,Hangup
exten=>6005,1,Dial(SIP/6005,30)
exten = 6005,n,Answer()
exten=>6005,n,Hangup

This is the new Log in *CLI>:

*CLI>        > Saved useragent "Zoiper rv2.9.2" for peer 6003
       > Saved useragent "Zoiper rv2.9.2" for peer 6001
sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
6001/6001                 xx.xx.xx.xx                            D  Auto (Yes) No             27370    Unmonitored                                  
6002/6002                 xx.xx.xx.120                           D  Auto (No)  No             1025     Unmonitored                                  
6003/6003                 xx.xx.xx.130                           D  Auto (Yes) No             54913    Unmonitored                                  
6004/6004                 xx.xx.xx.120                           D  Auto (No)  No             5060     Unmonitored                                  
6005/6005                 xx.xx.xx.94                            D  Auto (No)  No             5060     Unmonitored                                  
5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 0 offline]
*CLI>   == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
       > 0x7fc74400d070 -- Strict RTP learning after remote address set to: xx.xx.xx.xx:26762
    -- Executing [6004@from-internal:1] Dial("SIP/6001-00000000", "SIP/6004,30") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/6004
    -- Nobody picked up in 30000 ms
    -- Executing [6004@from-internal:2] Answer("SIP/6001-00000000", "") in new stack
       > 0x7fc74400d070 -- Strict RTP switching to RTP target address xx.xx.xx.xx:26762 as source
       > 0x7fc74400d070 -- Strict RTP learning complete - Locking on source address xx.xx.xx.xx:26762
    -- Executing [6004@from-internal:3] Hangup("SIP/6001-00000000", "") in new stack
  == Spawn extension (from-internal, 6004, 3) exited non-zero on 'SIP/6001-00000000'
[Feb 18 09:16:40] WARNING[19397]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 2d8fb4b4567bb8e06c666fb45c0b6157@xx.xx.xx43:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Feb 18 09:17:10] WARNING[19397]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission kb8gpSPvYhpXdiaCZ9u0Tg.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

*CLI>   == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
       > 0x7fc74400d040 -- Strict RTP learning after remote address set to: xx.xx.xx.xx:26762
    -- Executing [6005@from-internal:1] Dial("SIP/6001-00000002", "SIP/6005,30") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/6005
    -- Nobody picked up in 30000 ms
    -- Executing [6005@from-internal:2] Answer("SIP/6001-00000002", "") in new stack
       > 0x7fc74400d040 -- Strict RTP switching to RTP target address xx.xx.xx.xx:26762 as source
       > 0x7fc74400d040 -- Strict RTP learning complete - Locking on source address xx.xx.xx.xx:26762
    -- Executing [6005@from-internal:3] Hangup("SIP/6001-00000002", "") in new stack
  == Spawn extension (from-internal, 6005, 3) exited non-zero on 'SIP/6001-00000002'
[Feb 18 09:18:04] WARNING[19397]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 51cce20935cc88db0a46e9e8511dcc3c@xx.xx.xx43:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Feb 18 09:18:34] WARNING[19397]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission iae2tySEvqBoiDOEKYHi_Q.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response

You have not configured the device entries to know they are behind NAT. This is done using the “nat” option. Setting it to “nat=force_rport,comedia” may help.

Include this in sip.conf? and test again?

It can go into the general section or the individual sections.

Nothing works!
None softphone can comunicate with the other.
None softphone can comunicate with none camera, at least does not it hear nothing and there are not dial tone.
In cameras are auto answer setting up.
I don’t know, what to do! seem I’m going in circles.

New sip.conf file:

[general]
nat=force_rport,comedia
videosupport=yes
directmedia=no
context=public
disallow=all
allow = alaw,ulaw,g723,h263,gsm,g729
limitonpeers=yes
rtptimeout=600
session-timers=accept
session-expires=120
session-minse=90
session-refresher=uas
rtptimeout=600
notifyringing=notinuse
notifyhold=yes
callcounter=yes
allowsubscribe=yes
port=5060
bindaddr=0.0.0.0
language=es
[6001]
type=friend
context=from-internal
host=dynamic
secret=r****E
[6002]
type=friend
context=from-internal
host=dynamic
secret=r****E
[6003]
type=friend
context=from-internal
host=dynamic
secret=r****E
[6004]
type=friend
context=from-internal
host=dynamic
secret=r****E
[6005]
type=friend
context=from-internal
host=dynamic
secret=r****E

The Log of *CLI> with nothing working

*CLI>   == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
       > 0x7f26d0020a00 -- Strict RTP learning after remote address set to: xx.xx.xx.90:26762
    -- Executing [6005@from-internal:1] Dial("SIP/6001-00000002", "SIP/6005,30") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/6005
  == Spawn extension (from-internal, 6005, 1) exited non-zero on 'SIP/6001-00000002'
[Feb 18 09:44:36] WARNING[19847]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 495e99fe7a8b0ee30ecfedf009d90b85@64.157.15.43:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
       > 0x7f26d0020a00 -- Strict RTP learning after remote address set to: 192.168.1.83:25282
[Feb 18 09:44:38] NOTICE[19847][C-00000004]: chan_sip.c:26687 handle_request_invite: Call from '' (xx.xx.xx.48:63007) to extension '380638835858' rejected because extension not found in context 'public'.
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
       > 0x7f26d000a4a0 -- Strict RTP learning after remote address set to: xx.xx.xx.90:26762
    -- Executing [6002@from-internal:1] Dial("SIP/6001-00000004", "SIP/6002,30") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/6002
[Feb 18 09:45:10] WARNING[19847]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 1117122126-177559812-1048225286 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
    -- Nobody picked up in 30000 ms
    -- Executing [6002@from-internal:2] Answer("SIP/6001-00000004", "") in new stack
       > 0x7f26d000a4a0 -- Strict RTP switching to RTP target address xx.xx.xx.90:26762 as source
       > 0x7f26d000a4a0 -- Strict RTP learning complete - Locking on source address xx.xx.xx.90:26762
    -- Executing [6002@from-internal:3] Hangup("SIP/6001-00000004", "") in new stack
  == Spawn extension (from-internal, 6002, 3) exited non-zero on 'SIP/6001-00000004'
[Feb 18 09:45:12] WARNING[19847]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 7ef69a136a0f97df2e8b3d9752e98945@64.157.15.43:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

I would suggest providing the output of “sip set debug on” with a call attempt then so we can see what the SIP signaling actually contains.


Magically something works without touching anything.
But other stuff continues not working!

Here a lat *CLI> and attacht the debug file.

*CLI>        > Saved useragent "Zoiper rv2.9.2" for peer 6001
       > Saved useragent "Zoiper rv2.9.2" for peer 6003

*CLI> module reload logger
Module 'logger' reloaded successfully.
    -- Reloading module 'logger' (Logger)
 Asterisk Queue Logger restarted
*CLI>   == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
       > 0x7f2f680087f0 -- Strict RTP learning after remote address set to: 192.168.1.83:25282
[Feb 18 10:32:54] NOTICE[20472][C-00000001]: chan_sip.c:26687 handle_request_invite: Call from '' (46.8.xx.xx:63584) to extension '08011380638835858' rejected because extension not found in context 'public'.
[Feb 18 10:33:26] WARNING[20472]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission 463002703-1308614756-1734224443 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

*CLI> 
*CLI> Asterisk cleanly ending (0).

Thanks in advance for your support

debug_log_billyvb2004_02.txt (1.5 MB)