PJSIP : Nobody picked up

Asterisk Asterisk Support
1

Hi,

I’m new in Asterisk field. But I want to learn more about this great soft.

So I’ll explain the issue:

  • I’m working with Asterisk 13.3 and latest pjsip version.

  • I configured OpenVPN tunnel to pass audio calls through it . The tunnel works and clients can connect to OpenVPN server: and softphones show registered when connected to VPN server. (might be the cause)

  • When softphones register to Asterisk only these warnings are showing; I though that my problem is like this one "“https://forum.snom.com/index.php?showtopic=14573"” , but is the same when I added the patch as described there:

I get only this :

" Added contact ‘sip:6001@10.82.3.6:61506;rinstance=3ef662af83ebf79d’ to AOR ‘6001’ with expiration of 3600 seconds
[May 28 10:32:15] NOTICE[12893]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed"

And when softphones try to register : I get this :

“res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[May 28 11:17:49] WARNING[13537]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo”

But I really don’t know if this is a complete registration:
** “pjsip show registrations” command doesn’t show nothing despite softphones say registered.
** " pjsip list endpoints" command shows all endpoints but with “not in use” as status.

— The problem is when trying to make calls between endpoints: sometimes it works and many times don’t ; Asterisk showing only this :

“Executing [6001@from-internal:1] Dial(“PJSIP/6001-0000000a”, “PJSIP/6001,20”) in new stack
– Called PJSIP/6001
– Nobody picked up in 20000 ms
– Auto fallthrough, channel ‘PJSIP/6002-00000008’ status is ‘NOANSWER’”

And if it worked, when called endpoint “hang up the call”, the caller endpoint don’t figure out that (I mean that the call keeps running in caller side) !!!

Please have you any ideas, I have been blocked here for several days??? knowing that all was fine and worked when I tried with asterisk and endpoints in the same subnet with the same pjsip version.

2

Your problem is either that you are calling the same softphone from which you are making the call or you have multiple endpoints registered to the same AOR.

This message:

Indicates that your softphone successfully registered with an endpoint with AOR 6001. The command to dial this device is Dial(PJSIP/6001)

This message:

Means that you dont have a mailbox configured for this AOR. You can ignore this for now as it wont prevent you from making or receiving calls.

This patch:[b]https://forum.snom.com/index.php?showtopic=14573[/b] Is not related to these warnings.

You can ignore this.[quote]“res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[May 28 11:17:49] WARNING[13537]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo”[/quote]This is just the device trying to send a PUBLISH event that is not supported by Asterisk.

pjsip show registrations wont show because this commands lists outbound registrations and you are using inbound registrations, which is correct for registering softphones.

pjsip list endpoints is correct to say “Not in use” because that is the state of the phone when its not on a call or ringing.

This right here[quote]Executing [6001@from-internal:1] Dial(“PJSIP/6001-0000000a”, “PJSIP/6001,20”) in new stack
– Called PJSIP/6001
– Nobody picked up in 20000 ms
– Auto fallthrough, channel ‘PJSIP/6002-00000008’ status is ‘NOANSWER’"[/quote]This seems to say you are calling from PJSIP/6001 to PJSIP/6001. I’m not sure why there is the Auto fallthrough, channel ‘PJSIP/6002-00000008’ status is ‘NOANSWER’ You may be trying to register multiple endpoints to the same AOR? I’ll bet if you look closely at your softphone, it will show an incoming call while you are making this outbound call. This is like calling your cellphone from your cellphone, it doesnt work. What you need to do is register one phone with AOR 6001 and the other with AOR 6002. Post your pjsip.conf to be sure.

As an alternative, I’d recommend (I might be outvoted on this) starting with the old sip stack rather than pjsip. It is easier for beginners to learn and understand, in my opinion. Once you have sip figured out, you can look at how it translates to pjsip.

You should start with the O’Reily Asterisk book. Edition 3 is available for free online here asteriskdocs.org/ See the section on Online Versions in the middle of the page. It doesn’t cover chan_pjsip but it will cover chan_sip in enough detail to get you started and understand how Asterisk connects channels and devices.

3

The NOTICE is is occurring because the endpoint is trying to subscribe to receive voicemail information, but no mailboxes have been configured.

The WARNING is occurring because the endpoint is attempting to publish presence information but this is not currently supported.

As for your problem please turn on packet output using “pjsip set logger on” and provide the console output when an endpoint registers and when you call it and it fails.

4

Thanks for the reply,

For using PJSIP:
– When an endpoint registers I get only this :
"– Added contact ‘sip:6001@My.IP.IP.IP:14737;transport=UDP;rinstance=68516eaae9d9ff89’ to AOR ‘6001’ with expiration of 3600 seconds
[Jun 1 15:32:27] NOTICE[7757]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed
[Jun 1 15:32:27] WARNING[7758]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
"
– When try to make a call from 6003 extension to 6001extension, I get this:
"[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
[Jun 1 15:37:52] WARNING[7839]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
– Executing [6001@from-internal:1] Dial(“PJSIP/6003-00000002”, “PJSIP/6001,20”) in new stack
– Called PJSIP/6001
– Nobody picked up in 20000 ms
– Auto fallthrough, channel ‘PJSIP/6003-00000002’ status is ‘NOANSWER’
[Jun 1 15:38:13] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:38:13] WARNING[7839]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
[Jun 1 15:38:13] WARNING[7839]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Jun 1 15:38:13] WARNING[7840]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo"

==> This is a failed call.

Comparing to PJSIP, when using the old “chan_sip”, I get :
– When an extension registers, I get this :
" Registered SIP ‘6001’ at 10.82.3.1:45665
> Saved useragent “Z 3.3.25608 r25552” for peer 6001"

– When try to make a call from 6003 extension to 6001extension, I get this :
" == Using SIP RTP CoS mark 5
– Executing [6001@from-internal:1] Dial(“SIP/6003-00000004”, “SIP/6001,20”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/6001
– SIP/6001-00000005 is ringing
– SIP/6001-00000005 answered SIP/6003-00000004
– Channel SIP/6003-00000004 joined ‘simple_bridge’ basic-bridge
– Channel SIP/6001-00000005 joined ‘simple_bridge’ basic-bridge
> Bridge a2a0cb6d-ca53-43ba-949b-665417370969: switching from simple_bridge technology to native_rtp
> Remotely bridged ‘SIP/6001-00000005’ and ‘SIP/6003-00000004’ - media will flow directly between them
> Remotely bridged ‘SIP/6001-00000005’ and ‘SIP/6003-00000004’ - media will flow directly between them
– Started music on hold, class ‘default’, on channel ‘SIP/6001-00000005’
> 0x7faee40046c0 – Probation passed - setting RTP source address to 10.82.3.1:8002
> 0x7faee40046c0 – Probation passed - setting RTP source address to 10.82.3.1:8002
– Stopped music on hold on SIP/6001-00000005
– Channel SIP/6001-00000005 left ‘native_rtp’ basic-bridge
– Channel SIP/6003-00000004 left ‘native_rtp’ basic-bridge
== Spawn extension (from-internal, 6001, 1) exited non-zero on ‘SIP/6003-00000004’"

===> And of course using the old chan_sip never failed.

And thanks in advance.

5

Ah! yes it’s true one of my tests was to call 6001 to 6001.
– But I tried from/to different endpoints but the same , the problem it’s once the call can pass and often not .
– By the way with the old “chan_sip” as you said , it’s working perfectly.
– Also, I’m testing within within a VPN network which I set using OPEN VPN: so softphones are installed in clients computers (different from Asterisk machine): each has different IP.

For PJSIP.conf is :
"""""""""""""""""""""""""""
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.x.x.x/255.255.255.255

[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp-nat
auth=6001
aors=6001
direct_media=no
rtp_symmetric=yes
force_rport=yes

[6001]
type=auth
auth_type=userpass
password=xxxxxx
username=6001

[6001]
type=aor
max_contacts=2

[6003]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp-nat
auth=6003
aors=6003
direct_media=no
rtp_symmetric=yes
force_rport=yes

[6003]
type=auth
auth_type=userpass
password=xxxxxxx
username=6003

[6003]
type=aor
max_contacts=2
""""""""""""""""""""""""""""""""""""""""""""""""""""

6

You have not done “pjsip set logger on” and thus there is no SIP signaling present. There’s nothing else I can add to my previous reply until that is shown.

7

sorry for that, I enabled now :

– This is the log when 6003 endpoint registered from 10.82.3.6:

[code]<— Received SIP request (688 bytes) from UDP:10.82.3.6:36769 —>
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1—aae6fd08617f7759
Max-Forwards: 70
Contact: sip:6001@10.82.3.6:36769;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=2f3da15b
Call-ID: M0DJjW30HWu2EsecxsurAQ…
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

<— Transmitting SIP response (493 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—aae6fd08617f7759
Call-ID: M0DJjW30HWu2EsecxsurAQ…
From: sip:6001@10.82.3.1;tag=2f3da15b
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—aae6fd08617f7759
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229264/3699d0f42141836e0d749ddd4b93cec0”,opaque=“6c74d1a4699cbd39”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (982 bytes) from UDP:10.82.3.6:36769 —>
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1—ec490396e34fb232
Max-Forwards: 70
Contact: sip:6001@10.82.3.6:36769;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=2f3da15b
Call-ID: M0DJjW30HWu2EsecxsurAQ…
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1433229264/3699d0f42141836e0d749ddd4b93cec0”,uri=“sip:6001@10.82.3.1;transport=UDP”,response=“1469ffe1e59b71534a6c743df382de01”,cnonce=“e9cc7dff896e1d17ca52c89c71efe418”,nc=00000001,qop=auth,algorithm=md5,opaque="6c74d1a4699cbd39"
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:14:24] NOTICE[3882]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed
<— Transmitting SIP response (343 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—ec490396e34fb232
Call-ID: M0DJjW30HWu2EsecxsurAQ…
From: sip:6001@10.82.3.1;tag=2f3da15b
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—ec490396e34fb232
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (648 bytes) from UDP:10.82.3.6:36769 —>
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—1e24cd15822f1a35
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;rinstance=f87d40b39d8500af;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Allow-Events: presence, kpml
Content-Length: 0

<— Transmitting SIP response (496 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—1e24cd15822f1a35
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
From: sip:6001@10.82.3.1;tag=50473f16
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—1e24cd15822f1a35
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,opaque=“04f2cce67f4e0a18”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (937 bytes) from UDP:10.82.3.6:36769 —>
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—217f49534a0c019f
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;rinstance=f87d40b39d8500af;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,uri=“sip:10.82.3.1;transport=UDP”,response=“8e9035dea9cb6c1371ae0e36633d62f4”,cnonce=“4f98156ce965bb9d9792521edc895e28”,nc=00000001,qop=auth,algorithm=md5,opaque="04f2cce67f4e0a18"
Allow-Events: presence, kpml
Content-Length: 0

-- Added contact 'sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af' to AOR '6001' with expiration of 3600 seconds

<— Transmitting SIP response (486 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—217f49534a0c019f
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
From: sip:6001@10.82.3.1;tag=50473f16
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—217f49534a0c019f
CSeq: 2 REGISTER
Date: Tue, 02 Jun 2015 07:14:48 GMT
Contact: sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af;expires=3599
Expires: 3600
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (935 bytes) from UDP:10.82.3.6:36769 —>
PUBLISH sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—c0e9d546d0d4b7f7
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=7635ce49
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg…
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>



open
Online

<— Transmitting SIP response (495 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—c0e9d546d0d4b7f7
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg…
From: sip:6001@10.82.3.1;tag=7635ce49
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—c0e9d546d0d4b7f7
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,opaque=“2b05028c0fa8f159”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (687 bytes) from UDP:10.82.3.6:36769 —>
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—7eda2b41003d4900
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=920f5a46
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ…
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<— Transmitting SIP response (497 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—7eda2b41003d4900
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ…
From: sip:6001@10.82.3.1;tag=920f5a46
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—7eda2b41003d4900
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,opaque=“61d6d4d74040ebb4”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (1229 bytes) from UDP:10.82.3.6:36769 —>
PUBLISH sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—a25e216e831f79b3
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=7635ce49
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg…
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,uri=“sip:6001@10.82.3.1;transport=UDP”,response=“18c4dc04b29040f32f8191806e4fc99b”,cnonce=“b2cbbb25f1b6196f321c1a87d4a213c8”,nc=00000001,qop=auth,algorithm=md5,opaque="2b05028c0fa8f159"
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>



open
Online

[Jun 2 08:14:48] WARNING[3882]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
<— Transmitting SIP response (345 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—a25e216e831f79b3
Call-ID: ID4jdnY-bPyJ1Eu8u3uBVg…
From: sip:6001@10.82.3.1;tag=7635ce49
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—a25e216e831f79b3
CSeq: 2 PUBLISH
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (696 bytes) from UDP:10.82.3.6:36769 —>
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—db522e91d922bbfd
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=87708c2d
Call-ID: OqoWdMUgmKwWX79QbYRWgw…
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

<— Transmitting SIP response (497 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—db522e91d922bbfd
Call-ID: OqoWdMUgmKwWX79QbYRWgw…
From: sip:6001@10.82.3.1;tag=87708c2d
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—db522e91d922bbfd
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,opaque=“62dc3a765cff63e9”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (981 bytes) from UDP:10.82.3.6:36769 —>
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—fc55bb12bada1719
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=920f5a46
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ…
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,uri=“sip:6001@10.82.3.1;transport=UDP”,response=“503741850430785ca5a2cdb0a3a2803e”,cnonce=“4a86d18d8f6eb388d37fcddb7287218e”,nc=00000001,qop=auth,algorithm=md5,opaque="61d6d4d74040ebb4"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:14:48] WARNING[3882]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<— Transmitting SIP response (347 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—fc55bb12bada1719
Call-ID: PI5K6X2W0E_MmwMrk_LhUQ…
From: sip:6001@10.82.3.1;tag=920f5a46
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—fc55bb12bada1719
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (990 bytes) from UDP:10.82.3.6:36769 —>
SUBSCRIBE sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22741;branch=z9hG4bK-524287-1—94ebda0155275009
Max-Forwards: 70
Contact: sip:6001@41.227.185.41:22741;transport=UDP
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=87708c2d
Call-ID: OqoWdMUgmKwWX79QbYRWgw…
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,uri=“sip:6001@10.82.3.1;transport=UDP”,response=“28f3dd18a082f3941397f3aa6f7bd014”,cnonce=“528f98f4ffbf24005b79553a7fed6797”,nc=00000001,qop=auth,algorithm=md5,opaque="62dc3a765cff63e9"
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:14:48] NOTICE[3882]: res_pjsip_mwi.c:679 mwi_new_subscribe: AOR 6001 has no configured mailboxes. MWI subscription failed
<— Transmitting SIP response (347 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 41.227.185.41:22741;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—94ebda0155275009
Call-ID: OqoWdMUgmKwWX79QbYRWgw…
From: sip:6001@10.82.3.1;tag=87708c2d
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—94ebda0155275009
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0
[/code]

– This is when I try to call 6001 from 6003:

[code] – Executing [6001@from-internal:1] Dial(“PJSIP/6003-00000002”, “PJSIP/6001,20”) in new stack
– Called PJSIP/6001
<— Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: sip:6003@192.168.1.2;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: sip:6003@192.168.1.2;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: sip:6003@192.168.1.2;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: sip:6003@192.168.1.2;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (984 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj637cbe2e-65d6-473e-af34-262392012214
From: sip:6003@192.168.1.2;tag=0d40f5de-1785-4587-b09c-21931bf8c27b
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:f436ecb0-ba1c-4ff1-ab4e-ba3e54746837@41.227.246.48:5060
Call-ID: 8fedc553-3bd9-4c40-b3e4-d096e4735834
CSeq: 3370 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 235

v=0
o=- 1398057981 1398057981 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 10762 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: sip:6003@192.168.1.2;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: sip:6003@192.168.1.2;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Nobody picked up in 20000 ms
-- Auto fallthrough, channel 'PJSIP/6003-00000002' status is 'NOANSWER'

<— Transmitting SIP response (394 bytes) to UDP:10.82.3.1:52170 —>
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-ef571ce00d5612c5-1—d8754z-
Call-ID: ZmJkNWNhZThjMTI3NDZhMzkxY2Q0ODgzMTYwOWRkYzc.
From: sip:6003@10.82.3.1;tag=1bc1c118
To: sip:6001@10.82.3.1;tag=156db271-1bfe-4b85-bd03-967cb38c407a
CSeq: 2 INVITE
Server: Asterisk PBX 13.3.0
Reason: Q.850;cause=0
Content-Length: 0

<— Received SIP request (364 bytes) from UDP:10.82.3.1:52170 —>
ACK sip:6001@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-ef571ce00d5612c5-1—d8754z-
Max-Forwards: 70
To: sip:6001@10.82.3.1;tag=156db271-1bfe-4b85-bd03-967cb38c407a
From: sip:6003@10.82.3.1;transport=UDP;tag=1bc1c118
Call-ID: ZmJkNWNhZThjMTI3NDZhMzkxY2Q0ODgzMTYwOWRkYzc.
CSeq: 2 ACK
Content-Length: 0

<— Received SIP request (947 bytes) from UDP:10.82.3.1:52170 —>
PUBLISH sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-1d2ea0e06f9ddf6b-1—d8754z-
Max-Forwards: 70
Contact: sip:6003@41.227.185.41:22377;transport=UDP
To: sip:6003@10.82.3.1;transport=UDP
From: sip:6003@10.82.3.1;transport=UDP;tag=80d73a07
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>



open
Online

<— Received SIP request (699 bytes) from UDP:10.82.3.1:52170 —>
SUBSCRIBE sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-7dbade96a2574170-1—d8754z-
Max-Forwards: 70
Contact: sip:6003@41.227.185.41:22377;transport=UDP
To: sip:6003@10.82.3.1;transport=UDP
From: sip:6003@10.82.3.1;transport=UDP;tag=cc230f28
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<— Transmitting SIP response (531 bytes) to UDP:10.82.3.1:52170 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-1d2ea0e06f9ddf6b-1—d8754z-
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
From: sip:6003@10.82.3.1;tag=80d73a07
To: sip:6003@10.82.3.1;tag=z9hG4bK-d8754z-1d2ea0e06f9ddf6b-1—d8754z-
CSeq: 1 PUBLISH
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229420/95a537f084952af3a3ea68d64c087584”,opaque=“42a2d1f4470b23b1”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Transmitting SIP response (533 bytes) to UDP:10.82.3.1:52170 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-7dbade96a2574170-1—d8754z-
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
From: sip:6003@10.82.3.1;tag=cc230f28
To: sip:6003@10.82.3.1;tag=z9hG4bK-d8754z-7dbade96a2574170-1—d8754z-
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229420/95a537f084952af3a3ea68d64c087584”,opaque=“5bad92ba26acf5d2”,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (1241 bytes) from UDP:10.82.3.1:52170 —>
PUBLISH sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-865df2942948418d-1—d8754z-
Max-Forwards: 70
Contact: sip:6003@41.227.185.41:22377;transport=UDP
To: sip:6003@10.82.3.1;transport=UDP
From: sip:6003@10.82.3.1;transport=UDP;tag=80d73a07
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
CSeq: 2 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username=“6003”,realm=“asterisk”,nonce=“1433229420/95a537f084952af3a3ea68d64c087584”,uri=“sip:6003@10.82.3.1;transport=UDP”,response=“7d19d5516b062c4950af3b66e422daa8”,cnonce=“69e0be5ff1cedcd3f3fc1f6c4d2768b6”,nc=00000001,qop=auth,algorithm=md5,opaque="42a2d1f4470b23b1"
Event: presence
Allow-Events: presence, kpml
Content-Length: 257

<?xml version="1.0" encoding="UTF-8"?>



open
Online

<— Received SIP request (993 bytes) from UDP:10.82.3.1:52170 —>
SUBSCRIBE sip:6003@10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 41.227.185.41:22377;branch=z9hG4bK-d8754z-e84caa37b87f1926-1—d8754z-
Max-Forwards: 70
Contact: sip:6003@41.227.185.41:22377;transport=UDP
To: sip:6003@10.82.3.1;transport=UDP
From: sip:6003@10.82.3.1;transport=UDP;tag=cc230f28
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username=“6003”,realm=“asterisk”,nonce=“1433229420/95a537f084952af3a3ea68d64c087584”,uri=“sip:6003@10.82.3.1;transport=UDP”,response=“47585f59499796484eea24703badf14d”,cnonce=“cd9fcbea6e3890f70d61057bc85362f8”,nc=00000001,qop=auth,algorithm=md5,opaque="5bad92ba26acf5d2"
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

[Jun 2 08:17:00] WARNING[4028]: res_pjsip_pubsub.c:2870 pubsub_on_rx_publish_request: No registered publish handler for event presence
<— Transmitting SIP response (381 bytes) to UDP:10.82.3.1:52170 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-865df2942948418d-1—d8754z-
Call-ID: NGE3YzI0MWQ4OWQwY2YzMWRkZWY1NWJlNjE5N2NlMTU.
From: sip:6003@10.82.3.1;tag=80d73a07
To: sip:6003@10.82.3.1;tag=z9hG4bK-d8754z-865df2942948418d-1—d8754z-
CSeq: 2 PUBLISH
Server: Asterisk PBX 13.3.0
Content-Length: 0

[Jun 2 08:17:00] WARNING[4028]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
<— Transmitting SIP response (383 bytes) to UDP:10.82.3.1:52170 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 41.227.185.41:22377;rport=52170;received=10.82.3.1;branch=z9hG4bK-d8754z-e84caa37b87f1926-1—d8754z-
Call-ID: M2EwNWZkZDAzMzhlNzcwMDIwZTk0MmNlYmE4YmMxNDE.
From: sip:6003@10.82.3.1;tag=cc230f28
To: sip:6003@10.82.3.1;tag=z9hG4bK-d8754z-e84caa37b87f1926-1—d8754z-
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (928 bytes) from UDP:10.82.3.6:36769 —>
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1—f16af0e0002d56d8
Max-Forwards: 70
Contact: sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af;expires=0
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
CSeq: 3 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1433229288/af00ed84c0264288745ad15aba761c1f”,uri=“sip:10.82.3.1;transport=UDP”,response=“6780609ec66cbce6620932772d983410”,cnonce=“2a74c6f7e67c8f59db2b5d04ba4658b4”,nc=00000002,qop=auth,algorithm=md5,opaque="04f2cce67f4e0a18"
Allow-Events: presence, kpml
Content-Length: 0

<— Transmitting SIP response (503 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—f16af0e0002d56d8
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
From: sip:6001@10.82.3.1;tag=50473f16
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—f16af0e0002d56d8
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1433229429/a5a58229b5db5ac03171753460566da4”,opaque=“7fb07b9807a7f2ec”,stale=true,algorithm=md5,qop="auth"
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Received SIP request (928 bytes) from UDP:10.82.3.6:36769 —>
REGISTER sip:10.82.3.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.82.3.6:36769;branch=z9hG4bK-524287-1—1ed6831cf6a85375
Max-Forwards: 70
Contact: sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af;expires=0
To: sip:6001@10.82.3.1;transport=UDP
From: sip:6001@10.82.3.1;transport=UDP;tag=50473f16
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
CSeq: 4 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username=“6001”,realm=“asterisk”,nonce=“1433229429/a5a58229b5db5ac03171753460566da4”,uri=“sip:10.82.3.1;transport=UDP”,response=“ba95d4b1ecd561328befe44065529665”,cnonce=“a94932f2d5daa9657d6dea6bcf81c2d1”,nc=00000001,qop=auth,algorithm=md5,opaque="7fb07b9807a7f2ec"
Allow-Events: presence, kpml
Content-Length: 0

-- Attempted to remove non-existent contact 'sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af' from AOR '6001' by request

<— Transmitting SIP response (467 bytes) to UDP:10.82.3.6:36769 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.82.3.6:36769;rport=36769;received=10.82.3.6;branch=z9hG4bK-524287-1—1ed6831cf6a85375
Call-ID: 5U-0wgCX7Luyt5ZXzCybsg…
From: sip:6001@10.82.3.1;tag=50473f16
To: sip:6001@10.82.3.1;tag=z9hG4bK-524287-1—1ed6831cf6a85375
CSeq: 4 REGISTER
Date: Tue, 02 Jun 2015 07:17:09 GMT
Contact: sip:6001@41.227.246.48:22741;transport=UDP;rinstance=f87d40b39d8500af;expires=3458
Server: Asterisk PBX 13.3.0
Content-Length: 0

<— Transmitting SIP request (983 bytes) to UDP:41.227.185.41:22741 —>
INVITE sip:6001@41.227.185.41:22741;transport=UDP;rinstance=f87d40b39d8500af SIP/2.0
Via: SIP/2.0/UDP 41.227.246.48:5060;rport;branch=z9hG4bKPj6ee2ce59-4fdb-437f-b3f9-c7ce2edd884a
From: sip:6003@192.168.1.2;tag=ca401b00-ba52-438f-855a-0ad3741f0a42
To: sip:6001@41.227.185.41;rinstance=f87d40b39d8500af
Contact: sip:94097982-2a81-45b9-a46b-2e2b9033e30d@41.227.246.48:5060
Call-ID: a0035ee2-0e78-4efc-be32-fe6c1b81d7bf
CSeq: 20523 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 13.3.0
Content-Type: application/sdp
Content-Length: 233

v=0
o=- 732217806 732217806 IN IP4 192.168.1.2
s=Asterisk
c=IN IP4 192.168.1.2
t=0 0
m=audio 11766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[/code]

8

The register has specified a public address and a non-standard port number, but has come from a private address (and a different port number). I would look at the configuration of the peer and any firewall and NAT devices on the way.

9

Yes I’m using a private network which I configured using OPEN VPN to put all endpoints under the same subnet.
For the port:

  • I configured my router to be able to send/receive from the standard Asterisk ports ‘udp 5060’ and “10000 - 20000”.
  • I configured my firewall also to be able to receive/send from VPN interface.
10

The device registering as wanting outgoing calls to go to public address 41.227.185.41 and to the non-standard port number 22741.

Asterisk is correctly sending to that address (and giving its public address and the standard port number for responses). Either something is blocking traffic to 41.227.185.41:22741 or something Is blocking SIP traffic from that address to Asterisk, as Asterisk never sees any responses.

11

yes, you are right.

To be clear the public IP address you see is the external IP of Asterisk (and VPN server also): so normally all sip packets must be routed from/to this address.
And using VPN tunnel must affect all VPN clients : each must have unique IP within VPN subnet, but also must have the Server IP as front side ( in the front of external world): might be this is the problem ?

As I understand, the problem is between the External IP of the router------and Local IP of Asterisk: in this level we have only the router and the firewall. so :
----As I mentioned before, I set up the needed rules in the router : for 5060 port, 10000-20000 port to be routed to the local IP of Asterisk. (even when I tested I dropped down the router firewall to be certain : and same result).

---- so might be the problem is in my firewall rules: here is : can you give me your opinion ???
*** VPN rules (in rc.local file):

[code]# Allow TUN interface connections to OpenVPN server
iptables -A INPUT -i tun+ -j ACCEPT

Allow TUN interface connections to be forwarded through other interfaces

iptables -A FORWARD -i tun+ -j ACCEPT

Allow TAP interface connections to OpenVPN server

iptables -A INPUT -i tap+ -j ACCEPT

Allow TAP interface connections to be forwarded through other interfaces

iptables -A FORWARD -i tap+ -j ACCEPT

iptables -I INPUT 1 -p udp --dport xxxx -j ACCEPT
iptables -I INPUT 1 -p tcp --dport xxxx -j ACCEPT (xxxx=“default port of VPN”)

iptables -I FORWARD -i wlan0 -o tun0 -j ACCEPT
iptables -I FORWARD -i tun0 -o wlan0 -j ACCEPT
iptables -t nat -I POSTROUTING -o wlan0 -j MASQUERADE
iptables -t nat -A POSTROUTING -s 10.82.3.0/24 -j SNAT --to Asterisk.local.IP.IP[/code]

*** Asterisk rules:

iptables -I INPUT 11 -m udp -p udp --dport 5060 -j ACCEPT iptables -I INPUT 10 -m udp -p udp --dport 10000:20000 -j ACCEPT

Any ideas, I know there are extra rules (might be this is problem) but this what I did for testing: but it’s really strange since these are the same rules used when I tried with the old chan_sip (even with pjsip when sometimes it succeed in calling)!!

12

If you set rewrite_contact=yes in the endpoints, does that work?

13

Sorry for the delay,
– I tried with the old “Chan-sip”, and it works perfectly.

– Might be this not the context, but I need your opinion about this:
** My project is to set up asterisk for a call center, so I need to give in the end a working system with a GUI ( so a manager can supervise the system without coding and manipulating config files). For that I installed the “Old Asterisk-GUI”.
I really want to manipulate config files directly, so I can test any solution quickly. Also, Gui restrict files moodifications. For Example, Asterisk-Gui don’t support PJSIP channel to try “jcolp” solution for example. So I must every time I want to try something , I must uninstall the gui.

******my questions are :
–So Is there any way to make Asterisk-Gui consider my modifications on files (add sip account, add sip trunk …) ??? for example try to make all my implementation in Asterisk files then I try to import these configurations using this GUI ???

– My choice of "Old Asterisk-Gui " is it wise ??? Or I should change to another Gui : knowing that choose it because I believe that it is the most close Gui to “Asterisk environment” (Freepbx for example try to impose its coding templates on Asterisk files …) ???

– Or I should, develop a new GUI inspired from “Asterisk-GUI” to give a simple “Supervision Gui” to the final user in the same time I keep all control on Asterisk files ???

– Finally, If this is all not possible : can I find a way to make “Old Asterisk-GUI” use PJSIP channel instead of Chan-sip channel ?? And if I keep working with Chan-sip, what influences will be on Asterisk system with at least 400 simultaneous calls comparing to PJSIP use ???

And thanks.

14

Asterisk GUI has not been supported for several years and probably pre-dates PJSIP.

15

Yeah - it predates PJSIP by, oh, 5+ years? Probably longer. Time flies.

16

So should I switch to freepbx gui??
But in that case:
– I won’t be able to take control of my config files !
– I won’t be able to modify this GUI to show some features and hide some others, kind of privileges to GUI users .

Because, as I understand, “Asterisk-GUI” is open source and can be integrated.

17

FreePBX is open source. For both, you will be on your own if you modify the code, but for Asterisk GUI you will be on your own even with the unmodified product.

18

At least, from your experiences if I use Freepbx, can I change my config files freely (if I need it) ??? or I must do all from the GUI ?? because from my few manipulations on the old "Asterisk -GUI " I discovered that it’s possible here ( GUI take modifications in consideration).

Have you any ideas ?

19

I don’t use FreePBX. Normally any GUI expects total control, although FreePBX does allow some low level customisation, but you need to ask the FreePBX people about the details.

20

ok thanks. I think that I shuld take another look at “Freepbx” GUI, because problems start showing : can’t find or enable “Record audio call option” anywhere !!!