Sip call drops after 32 seconds

Sip call drops after 32 seconds DEFAULT

Solved - Calls disconnected after around 30 seconds

3CX - Software Based VoIP IP PBX / PABX

Solved - Calls disconnected after around 30 seconds

Dear all, the system used to work fine, but recently I'm having problems with external incoming calls getting disconnected after around 30 seconds. I have a ring group with three extensions, one extension (611) answers the call Activity log...

In very simple terms, ACK is a sip message that follows a 200 OK message to signify to the party that sends the 200 OK that this has been received. If there is no ACK received within 32 seconds then the call is dropped.

2017-09-18_14h23_03

This is a very basic call flow showing an incoming call to the PBX that is routed to an IP phone. As you can see after the initial Invites the call is answered by the IP phone and a 200 OK message is sent to the PBX. Then PBX then forwards the 200 OK message to the provider and sends an ACK to the phone. Once the provider receives the 200 OK from the PBX sends an ACK message to the PBX for confirmation.
This behaviour is specified in RFC3261.

You can run a capture and check where this is failing for you

3CX

3CX Phone System Build History / Change Log

Review all the changes made to all 3CX products such as 3CX Phone System, 3CXPhone, etc here. Build History updated from version 6.0.366 to current version.

YiannisH_3CX

YiannisH_3CX

Support Team

Staff member

3CX Support

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Sep 26, 2017

I don’t think that it was the PBX delivers audio that temporarily solved the issue as the audio would still be routed through the PBX for local extensions. You mentioned that the issue is only with extensions in the other end of the VPN. To properly troubleshoot the issue you need to run a wireshark capture on the PBX and on the phone through the web interface of the device. Once you start both captures make a call and wait for it to drop. Save the captures and send them to me in a p.m so i can tale a look. This way we will be able to see what happens in both ends of the VPN and perhaps find the reason this happens. Before doing so please enable the PBX delivers audio option under the trunk settings.

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narkumas

narkumas

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Sep 29, 2017

Hey Yiannis!
Thank You for the superior support. The issue is fixed now. It works like a charm!
You can close this.

In case someone will have the same problem in the future:
Reason for the disconnect after 32 seconds was an old template for my QSC trunks. The Trunk-Hostname was wrong.
It pointed to a host with TCP connection. Disconnection happened when ACK was not delivered.
With UDP there certainly is no such problem.

Sours: https://hostcheetah.com/t/solved-calls-disconnected-after-around-30-seconds/675

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drop sip calls after 32 seconds

drop sip calls after 32 seconds

mode1(Programmer)

(OP)

I have been battling this for awhile at a customer site. Site has IP office R9.1.7. I have the same setup at my office using same sip provider and same release of ip office with no trouble. The sip provider recently changed to a new peering sip server. I believe this is the trouble. I pointed my customer's sip trunks to my office and internet and my sip trunks work and my customer does the same thing with the drop after 32 seconds. I called the provider and they did not have a reason why. They say they see back and forth 200 messages then a bye message. I have attached a monitor trace of the dropped call. Any help would be greatly appreciated.

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Hi, we were running v16 update 4 with no issues. Our pbx is on-prem running on a hyperv host. After updating to update 5 for the new app, our calls are disconnecting after 32 seconds. Error is "ACK is not received". Firewall checker shows all green. our firewall shows nothing dropped. I've updated before with no issues so wondering what could be happening. 

From what I've been reading, this points to a firewall issue, but I rolled back the update with a snapshot and everything is working normally, no changes to firewall

Anyone have any ideas? Thanks.


5 Replies

· · ·

JoeWilliams

Mace

OP

It certainly *sounds* firewally.

First check - does it affect internal calls?

Has anything changed in the SIP trunk settings for your provider as part of the update?

You might need to packet capture before & after calls and analyse them with Wireshark to see where the difference is.

0

· · ·

glenknox

Pimiento

OP

Well, when I first tried the update, I was working remotely. So I'm heading in tomorrow to try again and I'll be able to test internally. Thanks

0

· · ·

Pete ITProFL

Datil

OP

Does the firewall checker pass? Seems like a port forwarding issue to me, so usually firewall/NaT related. Usually when I see the 30 second disconnect it’s stun related but you said on prem. Are you sure the phones are connecting on LAN and not STUN?

0

· · ·

glenknox

Pimiento

OP

Ok, found the problem, it was the firewall! the 3CX firewall checker passed with no issues. So, updating 3CX is like a re-install, I just kept clicking "next, next, next". I found out that it was pulling our default public IP (x.x.x.170) instead of the IP for our phone system (x.x.x.172). So of course our firewall was dropping it. I've updated in the past and don't remember having to change anything during the re-setup, I guess I'll just have to keep a closer eye next time. Hope this helps someone. Thanks all!

2

· · ·

JoeWilliams

Mace

OP

glenknox wrote:

Ok, found the problem, it was the firewall! the 3CX firewall checker passed with no issues. So, updating 3CX is like a re-install, I just kept clicking "next, next, next". I found out that it was pulling our default public IP (x.x.x.170) instead of the IP for our phone system (x.x.x.172). So of course our firewall was dropping it. I've updated in the past and don't remember having to change anything during the re-setup, I guess I'll just have to keep a closer eye next time. Hope this helps someone. Thanks all!

Interestingly, I had exactly the same issue with the latest update :)

0

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Sours: https://community.spiceworks.com/topic/2276566-3cx-dropping-calls-after-30-seconds-after-update

All SIP phone calls drop after 32 seconds exactly

Is this sufficient Tom? It captured a lot of traffic so I hope I grabbed enough for you?

Content-Length: 296

v=0
o=- 6666 5 IN IP4 192.168.0.19
s=Asterisk
c=IN IP4 192.168.0.19
t=0 0
m=audio 11356 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2019-02-08 15:00:41] VERBOSE[5379] res_pjsip_logger.c: <— Transmitting SIP request (483 bytes) to UDP:192.168.0.58:59273 —>
OPTIONS sip:[email protected]:59273;rinstance=f13350a4f8e8b87a SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPj68fe4556-cb19-4fed-b89d-7c0a3fd76c81
From: sip:[email protected];tag=0fb1151a-69d8-4a4d-911b-bac605a54ce4
To: sip:[email protected];rinstance=f13350a4f8e8b87a
Contact: sip:[email protected]:5060
Call-ID: 9d0ab605-7c3e-4820-a3e6-e320ea1cd57f
CSeq: 13165 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-02-08 15:00:41] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (627 bytes) from UDP:192.168.0.58:59273 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPj68fe4556-cb19-4fed-b89d-7c0a3fd76c81;received=192.168.0.19
Contact: sip:192.168.0.58:59273
To: sip:[email protected];rinstance=f13350a4f8e8b87a;tag=b8a9095a
From: sip:[email protected];tag=0fb1151a-69d8-4a4d-911b-bac605a54ce4
Call-ID: 9d0ab605-7c3e-4820-a3e6-e320ea1cd57f
CSeq: 13165 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Allow-Events: talk, hold
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] app_dial.c: DAHDI/4-1 answered DAHDI/1-1
[2019-02-08 15:00:43] VERBOSE[9190] res_pjsip_logger.c: <— Transmitting SIP request (804 bytes) to UDP:10.8.0.17:5060 —>
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.1:5060;rport;branch=z9hG4bKPj0b3973b2-e52c-4479-8038-1fe82a3a42a9
From: sip:[email protected];tag=97e602fe-85ca-46f9-908e-da3c2a02c4ef
To: “E-Home” sip:[email protected];tag=1ad2846ee1c0d01
Contact: sip:10.8.0.1:5060
Call-ID: [email protected]
CSeq: 29812 NOTIFY
Event: dialog
Subscription-State: active;expires=261
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Type: application/dialog-info+xml
Content-Length: 231

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

[2019-02-08 15:00:43] VERBOSE[9190] res_pjsip_logger.c: <— Transmitting SIP request (827 bytes) to UDP:192.168.0.62:5060 —>
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPj22c4b144-0a99-4537-a60a-2b8590f726b9
From: sip:[email protected];tag=bec3e079-b096-47d8-8ea3-76d1675ad92c
To: “Cindy” sip:[email protected];tag=779a34eea22151d
Contact: sip:my.wan.ip.addr:5060
Call-ID: [email protected]
CSeq: 19540 NOTIFY
Event: dialog
Subscription-State: active;expires=54
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Type: application/dialog-info+xml
Content-Length: 230

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

[2019-02-08 15:00:43] VERBOSE[5379] res_pjsip_logger.c: <— Transmitting SIP request (483 bytes) to UDP:192.168.0.62:5160 —>
CANCEL sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPj2f3c9dcb-8490-4dbe-9806-9d0d614bda6e
From: “Nelson, BC” sip:[email protected];tag=3ad7b6a5-8289-4142-b418-5409370b9213
To: sip:[email protected]
Call-ID: 7241848e-7247-4097-827f-5db2a3a745f3
CSeq: 14560 CANCEL
Reason: SIP;cause=200;text=“Call completed elsewhere”
Reason: Q.850;cause=26
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[7166] res_pjsip_logger.c: <— Transmitting SIP request (804 bytes) to UDP:10.8.0.17:5060 —>
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.1:5060;rport;branch=z9hG4bKPj96a4181e-6cb7-4928-88e3-91e1845307c8
From: sip:[email protected];tag=42e62f0d-3a80-4d16-b54c-20534c8d1618
To: “E-Home” sip:[email protected];tag=83810dfb64c6303
Contact: sip:10.8.0.1:5060
Call-ID: [email protected]
CSeq: 17036 NOTIFY
Event: dialog
Subscription-State: active;expires=263
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Type: application/dialog-info+xml
Content-Length: 231

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

[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:1] Set(“DAHDI/4-1”, “__MACRO_RESULT=”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:2] Set(“DAHDI/4-1”, “CFIGNORE=”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:3] Set(“DAHDI/4-1”, “MASTER_CHANNEL(CFIGNORE)=”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:4] Set(“DAHDI/4-1”, “FORWARD_CONTEXT=from-internal”) in new stack
[2019-02-08 15:00:43] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (483 bytes) from UDP:192.168.0.62:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPj22c4b144-0a99-4537-a60a-2b8590f726b9;received=192.168.0.19
From: sip:[email protected];tag=bec3e079-b096-47d8-8ea3-76d1675ad92c
To: “Cindy” sip:[email protected];tag=779a34eea22151d
Call-ID: [email protected]
CSeq: 19540 NOTIFY
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:5] Set(“DAHDI/4-1”, “MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:6] Macro(“DAHDI/4-1”, “blkvm-clr,”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:1] Set(“DAHDI/4-1”, “SHARED(BLKVM,DAHDI/1-1)=”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:2] Set(“DAHDI/4-1”, “GOSUB_RETVAL=”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:3] MacroExit(“DAHDI/4-1”, “”) in new stack
[2019-02-08 15:00:43] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (517 bytes) from UDP:192.168.0.62:5160 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPj2f3c9dcb-8490-4dbe-9806-9d0d614bda6e;received=192.168.0.19
From: “Nelson, BC” sip:[email protected];tag=3ad7b6a5-8289-4142-b418-5409370b9213
To: sip:[email protected];tag=50142df6e6547d5
Call-ID: 7241848e-7247-4097-827f-5db2a3a745f3
CSeq: 14560 INVITE
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[5379] res_pjsip_logger.c: <— Transmitting SIP request (418 bytes) to UDP:192.168.0.62:5160 —>
ACK sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPj2f3c9dcb-8490-4dbe-9806-9d0d614bda6e
From: “Nelson, BC” sip:[email protected];tag=3ad7b6a5-8289-4142-b418-5409370b9213
To: sip:[email protected];tag=50142df6e6547d5
Call-ID: 7241848e-7247-4097-827f-5db2a3a745f3
CSeq: 14560 ACK
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:7] ExecIf(“DAHDI/4-1”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=4)”) in new stack
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] pbx.c: Executing [[email protected]:8] ExecIf(“DAHDI/4-1”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)”) in new stack
[2019-02-08 15:00:43] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (502 bytes) from UDP:192.168.0.62:5160 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPj2f3c9dcb-8490-4dbe-9806-9d0d614bda6e;received=192.168.0.19
From: “Nelson, BC” sip:[email protected];tag=3ad7b6a5-8289-4142-b418-5409370b9213
To: sip:[email protected];tag=50142df6e6547d5
Call-ID: 7241848e-7247-4097-827f-5db2a3a745f3
CSeq: 14560 CANCEL
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[18209][C-00000039] bridge_channel.c: Channel DAHDI/4-1 joined ‘simple_bridge’ basic-bridge <9e2f9585-2c3d-4c41-a95a-ab1e32ba653d>
[2019-02-08 15:00:43] VERBOSE[18206][C-00000039] bridge_channel.c: Channel DAHDI/1-1 joined ‘simple_bridge’ basic-bridge <9e2f9585-2c3d-4c41-a95a-ab1e32ba653d>
[2019-02-08 15:00:43] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (445 bytes) from UDP:10.8.0.17:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.0.1:5060;rport=5060;branch=z9hG4bKPj0b3973b2-e52c-4479-8038-1fe82a3a42a9
From: sip:[email protected];tag=97e602fe-85ca-46f9-908e-da3c2a02c4ef
To: “E-Home” sip:[email protected];tag=1ad2846ee1c0d01
Call-ID: [email protected]
CSeq: 29812 NOTIFY
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (445 bytes) from UDP:10.8.0.17:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.0.1:5060;rport=5060;branch=z9hG4bKPj96a4181e-6cb7-4928-88e3-91e1845307c8
From: sip:[email protected];tag=42e62f0d-3a80-4d16-b54c-20534c8d1618
To: “E-Home” sip:[email protected];tag=83810dfb64c6303
Call-ID: [email protected]
CSeq: 17036 NOTIFY
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:43] VERBOSE[2230] res_pjsip_logger.c: <— Transmitting SIP response (937 bytes) to UDP:192.168.0.73:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.73:5060;rport=5060;received=192.168.0.73;branch=z9hG4bKda9e3d80
Call-ID: [email protected]
From: “Edwin” sip:[email protected];tag=a36d5635eb51bfa
To: sip:[email protected];tag=b01321bc-4b03-42a0-b704-42f4771a159e
CSeq: 21 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: sip:my.wan.ip.addr:5060
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: “3195(Available)” sip:[email protected]
Content-Type: application/sdp
Content-Length: 296

v=0
o=- 6666 5 IN IP4 192.168.0.19
s=Asterisk
c=IN IP4 192.168.0.19
t=0 0
m=audio 11356 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2019-02-08 15:00:47] VERBOSE[2230] res_pjsip_logger.c: <— Transmitting SIP response (937 bytes) to UDP:192.168.0.73:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.73:5060;rport=5060;received=192.168.0.73;branch=z9hG4bKda9e3d80
Call-ID: [email protected]
From: “Edwin” sip:[email protected];tag=a36d5635eb51bfa
To: sip:[email protected];tag=b01321bc-4b03-42a0-b704-42f4771a159e
CSeq: 21 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: sip:my.wan.ip.addr:5060
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: “3195(Available)” sip:[email protected]
Content-Type: application/sdp
Content-Length: 296

v=0
o=- 6666 5 IN IP4 192.168.0.19
s=Asterisk
c=IN IP4 192.168.0.19
t=0 0
m=audio 11356 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2019-02-08 15:00:51] VERBOSE[2230] res_pjsip_logger.c: <— Transmitting SIP response (937 bytes) to UDP:192.168.0.73:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.73:5060;rport=5060;received=192.168.0.73;branch=z9hG4bKda9e3d80
Call-ID: [email protected]
From: “Edwin” sip:[email protected];tag=a36d5635eb51bfa
To: sip:[email protected];tag=b01321bc-4b03-42a0-b704-42f4771a159e
CSeq: 21 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: sip:my.wan.ip.addr:5060
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: “3195(Available)” sip:[email protected]
Content-Type: application/sdp
Content-Length: 296

v=0
o=- 6666 5 IN IP4 192.168.0.19
s=Asterisk
c=IN IP4 192.168.0.19
t=0 0
m=audio 11356 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2019-02-08 15:00:55] VERBOSE[2230] res_pjsip_logger.c: <— Transmitting SIP response (937 bytes) to UDP:192.168.0.73:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.73:5060;rport=5060;received=192.168.0.73;branch=z9hG4bKda9e3d80
Call-ID: [email protected]
From: “Edwin” sip:[email protected];tag=a36d5635eb51bfa
To: sip:[email protected];tag=b01321bc-4b03-42a0-b704-42f4771a159e
CSeq: 21 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: sip:my.wan.ip.addr:5060
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: “3195(Available)” sip:[email protected]
Content-Type: application/sdp
Content-Length: 296

v=0
o=- 6666 5 IN IP4 192.168.0.19
s=Asterisk
c=IN IP4 192.168.0.19
t=0 0
m=audio 11356 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2019-02-08 15:00:56] VERBOSE[2230] res_pjsip_logger.c: <— Transmitting SIP request (398 bytes) to UDP:192.168.0.73:5060 —>
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPjba53d711-7cb8-4169-b833-c7ef932302e1
From: sip:[email protected];tag=b01321bc-4b03-42a0-b704-42f4771a159e
To: “Edwin” sip:[email protected];tag=a36d5635eb51bfa
Call-ID: [email protected]
CSeq: 2233 BYE
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-02-08 15:00:56] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (479 bytes) from UDP:192.168.0.73:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPjba53d711-7cb8-4169-b833-c7ef932302e1;received=192.168.0.19
From: sip:[email protected];tag=b01321bc-4b03-42a0-b704-42f4771a159e
To: “Edwin” sip:[email protected];tag=a36d5635eb51bfa
Call-ID: [email protected]
CSeq: 2233 BYE
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] bridge_channel.c: Channel PJSIP/3160-0000002e left ‘simple_bridge’ basic-bridge
[2019-02-08 15:00:56] VERBOSE[18205][C-00000038] bridge_channel.c: Channel PJSIP/3195-0000002f left ‘simple_bridge’ basic-bridge
[2019-02-08 15:00:56] VERBOSE[7166] res_pjsip_logger.c: <— Transmitting SIP request (452 bytes) to UDP:192.168.0.58:59273 —>
BYE sip:[email protected]:59273 SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPj21ef00f1-6a01-44e1-a8f7-be0fda3a1ee3
From: “Edwin” sip:[email protected];tag=d80abe5e-66cc-4425-907a-b79acec72df8
To: sip:[email protected];rinstance=f13350a4f8e8b87a;tag=861c774f
Call-ID: fdb15862-49ef-4103-94d9-66d0705b6ef9
CSeq: 14775 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] app_macro.c: Spawn extension (macro-dial-one, s, 54) exited non-zero on ‘PJSIP/3160-0000002e’ in macro ‘dial-one’
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] app_macro.c: Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘PJSIP/3160-0000002e’ in macro ‘exten-vm’
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Spawn extension (ext-local, 3195, 2) exited non-zero on ‘PJSIP/3160-0000002e’
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/3160-0000002e”, “hangupcall,”) in new stack
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/3160-0000002e”, “1?theend”) in new stack
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/3160-0000002e”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/3160-0000002e”, "PJSIP/3195-0000002f montior file= ") in new stack
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Executing [[email protected]:5] GotoIf(“PJSIP/3160-0000002e”, “1?skipagi”) in new stack
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Executing [[email protected]:7] Hangup(“PJSIP/3160-0000002e”, “”) in new stack
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/3160-0000002e’ in macro ‘hangupcall’
[2019-02-08 15:00:56] VERBOSE[18166][C-00000038] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/3160-0000002e’
[2019-02-08 15:00:56] VERBOSE[9190] res_pjsip_logger.c: <— Transmitting SIP request (828 bytes) to UDP:192.168.0.62:5060 —>
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPj18fbc0fc-7fee-4941-ba1f-c2cc98658e19
From: sip:[email protected];tag=2966c454-2d42-49f2-91c6-dad2d78c6046
To: “Cindy” sip:[email protected];tag=b25dfc03ae1d540
Contact: sip:my.wan.ip.addr:5060
Call-ID: [email protected]
CSeq: 27231 NOTIFY
Event: dialog
Subscription-State: active;expires=41
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Type: application/dialog-info+xml
Content-Length: 231

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

[2019-02-08 15:00:56] VERBOSE[7166] res_pjsip_logger.c: <— Transmitting SIP request (805 bytes) to UDP:10.8.0.17:5060 —>
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.1:5060;rport;branch=z9hG4bKPjf03e5614-94e0-49b6-9262-0af2b5bf7d2d
From: sip:[email protected];tag=8dbe3380-0583-4b6e-92c6-9306592d5f5e
To: “E-Home” sip:[email protected];tag=b7080b58ad68174
Contact: sip:10.8.0.1:5060
Call-ID: [email protected]
CSeq: 17118 NOTIFY
Event: dialog
Subscription-State: active;expires=250
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Type: application/dialog-info+xml
Content-Length: 232

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

[2019-02-08 15:00:56] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (483 bytes) from UDP:192.168.0.62:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPj18fbc0fc-7fee-4941-ba1f-c2cc98658e19;received=192.168.0.19
From: sip:[email protected];tag=2966c454-2d42-49f2-91c6-dad2d78c6046
To: “Cindy” sip:[email protected];tag=b25dfc03ae1d540
Call-ID: [email protected]
CSeq: 27231 NOTIFY
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:56] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (445 bytes) from UDP:10.8.0.17:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.0.1:5060;rport=5060;branch=z9hG4bKPjf03e5614-94e0-49b6-9262-0af2b5bf7d2d
From: sip:[email protected];tag=8dbe3380-0583-4b6e-92c6-9306592d5f5e
To: “E-Home” sip:[email protected];tag=b7080b58ad68174
Call-ID: [email protected]
CSeq: 17118 NOTIFY
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0

[2019-02-08 15:00:56] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (460 bytes) from UDP:192.168.0.58:59273 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPj21ef00f1-6a01-44e1-a8f7-be0fda3a1ee3;received=192.168.0.19
Contact: sip:[email protected]:59273
To: sip:[email protected];rinstance=f13350a4f8e8b87a;tag=861c774f
From: “Edwin” sip:[email protected];tag=d80abe5e-66cc-4425-907a-b79acec72df8
Call-ID: fdb15862-49ef-4103-94d9-66d0705b6ef9
CSeq: 14775 BYE
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 0

[2019-02-08 15:00:58] VERBOSE[5379] res_pjsip_logger.c: <— Transmitting SIP request (428 bytes) to UDP:192.168.0.74:5160 —>
OPTIONS sip:[email protected]:5160 SIP/2.0
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport;branch=z9hG4bKPja20ce31c-2798-4199-94b4-217cd28da511
From: sip:[email protected];tag=485ab16a-26ca-4cf0-8ac9-7044febb8610
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: 56a99958-47d0-4c71-a0ef-4c27e3c03860
CSeq: 35176 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-02-08 15:00:58] VERBOSE[2230] res_pjsip_logger.c: <— Received SIP response (484 bytes) from UDP:192.168.0.74:5160 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.wan.ip.addr:5060;rport=5060;branch=z9hG4bKPja20ce31c-2798-4199-94b4-217cd28da511;received=192.168.0.19
From: sip:[email protected];tag=485ab16a-26ca-4cf0-8ac9-7044febb8610
To: sip:[email protected];tag=4f0e98b2e1d4cae
Call-ID: 56a99958-47d0-4c71-a0ef-4c27e3c03860
CSeq: 35176 OPTIONS
User-Agent: Sangoma S500 V2.0.4.57
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, PRACK
Content-Length: 0


Sours: https://community.freepbx.org/t/all-sip-phone-calls-drop-after-32-seconds-exactly/56306

After drops 32 seconds sip call

Freeswitch drops calls after 32 seconds

So I've installed Freeswitch on a raspberry PI 3 and it's dropping calls after 32 seconds. I've googled extensively and this appears to be a common problem but all of the people with the problem had complicated setups with external gateways, VPNs, NAT, multiple subnets etc. In my case I'm using almost bog stock config, the only change I've made is to delete the IPv6 profiles from the config. I haven't even changed the default password. I'm using everything on a home network with a /24 subnet and all phones internal. At first I thought it was the network dropping packets but it only happens with some phones. The 2 phones are the android version of Zoiper and an older version of Zoiper on my PC. I have the newer version of Zoiper on the same PC and no problems. Everything I find refers to NAT, STUN, VPNs, firewalls etc. I have none of that, firewall is disabled on the PC and the PI. Traffic isn't going through my router. One instance that isn't working is using UDP, the other TCP. All the working ones are TCP. This is bog stock out the box. Any ideas?

asked Mar 27 '20 at 10:40

MikeKullsMikeKulls

28711 gold badge22 silver badges1111 bronze badges

Sours: https://serverfault.com/questions/1008661/freeswitch-drops-calls-after-32-seconds
Short Explanation - SIP Signaling - Basic Call Flow - SIP

Then tea and sweet eclairs arrived, and the conversation became much warmer. She has an ambush with money, she has to go home to her mother, but there is no money for a ticket, DEZ does. Not pay, and neither does the tenants. In general, everything is bad.

She is clearly embarrassed and hardly looks into my eyes.

Now discussing:

"He asked. The girl shrank even more in response and nodded her head. The doctor approached her and with his hand tightened in a rubber glove, gently parted the elastic female buttocks, and smoothly introduced the long tip into the patient's ass.



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