Pjsip.conf flowroute config assistance - no response

Asterisk Asterisk SIP
1

I built a dedicated box on Fedora for my PBX. It seems to have run flawlessly until a few months ago.

Now it has an intermittent problem where Flowroute says my PBX is not responding. Other times everything seems to work.

I’m happy to provide more config detail and tcpdump data–but thought posting my config, 1st, might be a good starting point. I built this following directions to migrate from sip.conf to pjsip.conf.

Is there anything wrong with the following config?

==============================================================================================
; UDP transport behind NAT
;
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.1.0/24
external_media_address=155.155.155.155     ; my pub IP addr
external_signaling_address=155.155.155.155 ; my pub IP addr
;external_signaling_port=5060 # default
===================================================================================
[flowroute]
type=registration
transport=transport-udp-nat 
outbound_auth=flowroute_auth
server_uri=sip:us-east-va.sip.flowroute.com
client_uri=sip:[tech prefix]@us-east-va.sip.flowroute.com
contact_user=[tech prefix]
retry_interval=60      
;forbidden_retry_interval=600
;expiration=3600        
;line=yes
;endpoint=flowroute 

[flowroute_auth]
type=auth
auth_type=userpass
password=mypassword
username=[tech prefix]
realm=sip.flowroute.com

[flowroute]
type=aor
contact=sip:us-east-va.sip.flowroute.com

[flowroute]
type=endpoint
context=from-trunk
disallow=all
allow=ulaw
outbound_auth=flowroute_auth
aors=flowroute
direct_media=no

[flowroute]
type=identify
endpoint=flowroute
match=34.210.91.112/28,34.226.36.32/28,16.163.86.112/30,3.0.5.12/30,3.8.37.20/30,3.71.103.56/30,18.228.70.48/30,216.115.69.144

Thank you very much, in advance, for any advice and/or pointers!

2

Please provide full logs with “pjsip set logger on” enabled, showing a transaction that Flowroute claim to have failed.

3

Naturally, it’s been working since your post. I have pjsip set logger on set.

When it next fails (1-hour to 3 days) I’ll post.

Thank you.

4

you may want to use an external SIP debug tool, to make it easyer to look though several calls / trafic

  • tcpdump
  • sngrep
  • homer
5

I have:

pjsip set logger on
core set verbose 5
core set debug 5

I tried this evening and noticed I was experiencing the issue. I looked in /var/log/asterisk/messages and the log has only one line.

What’s the best thing I can do to get to the point where I can provide the necessary log?

@TheMark - I have been running a cap using the following:

~]# tcpdump -v net 34.226.36.32/28 or net 34.226.36.32/28 > /root/freenas/watchtcpip$(date "+%y%m%d-%H-%M-%S").txt

This grabs flowroute’s us-east and us-west networks. I could provide the entire log after I remove the private detail.

Thanks!

6

OK, it’s failing. I don’t see any log updates (as previously described).

Here’s a snippet from my tcpdump (as previously described)–does this help?

13:37:24.559335 IP (tos 0x0, ttl 52, id 17761, offset 0, flags [none], proto UDP (17), length 559)
    ec2-34-226-36-35.compute-1.amazonaws.com.sip > myPBX.sip: SIP, length: 531
        OPTIONS sip:[techprefix]@[my ip addr]:5060 SIP/2.0
        Max-Forwards: 20
        Record-Route: <sip:34.226.36.35;lr>
        Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bK69a1.d7dabe6a5d7f81cb0f12b9296ee7dc47.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK8932332
        Route: <sip:34.226.36.35:5060;lr;received=sip:[my ip addr]:5060>
        From: sip:ping@invalid;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-66b5dc77
        To: sip:[techprefix]@[my ip addr]:5060
        Call-ID: 75694be1-cbb3dd09-1ac5fe8@0.0.0.0
        CSeq: 1 OPTIONS
        Content-Length: 0
        Max-Forward: 10
        
13:37:24.559943 IP (tos 0x0, ttl 64, id 34302, offset 0, flags [DF], proto UDP (17), length 1027)
    myPBX.sip > ec2-34-226-36-35.compute-1.amazonaws.com.sip: SIP, length: 999
        SIP/2.0 404 Not Found
        Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bK69a1.d7dabe6a5d7f81cb0f12b9296ee7dc47.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK8932332
        Record-Route: <sip:34.226.36.35;lr>
        Call-ID: 75694be1-cbb3dd09-1ac5fe8@0.0.0.0
        From: <sip:ping@invalid>;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-66b5dc77
        To: <sip:[techprefix]@[my ap addr]>;tag=z9hG4bK69a1.d7dabe6a5d7f81cb0f12b9296ee7dc47.0
        CSeq: 1 OPTIONS
        Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary
, application/sdp, message/sipfrag;version=2.0
        Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
        Supported: 100rel, timer, replaces, norefersub
        Accept-Encoding: identity
        Accept-Language: en
        Server: Asterisk PBX 18.4.0
        Content-Length:  0
        
13:38:19.803338 IP (tos 0x0, ttl 52, id 23332, offset 0, flags [none], proto UDP (17), length 559)
    ec2-34-226-36-35.compute-1.amazonaws.com.sip > myPBX.sip: SIP, length: 531
        OPTIONS sip:[techprefix]@[my ip addr]:5060 SIP/2.0
        Max-Forwards: 20
        Record-Route: <sip:34.226.36.35;lr>
        Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bK3438.24c2c0a277d1fe3eef03b45aaf36f737.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK9414134
        Route: <sip:34.226.36.35:5060;lr;received=sip:[my ip addr]:5060>
        From: sip:ping@invalid;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-4438dc77
        To: [techprefix]@[my ap addr]:5060
        Call-ID: 75694be1-a936dd09-8dc5fe8@0.0.0.0
        CSeq: 1 OPTIONS
        Content-Length: 0
        Max-Forward: 10
        
13:38:19.803999 IP (tos 0x0, ttl 64, id 54707, offset 0, flags [DF], proto UDP (17), length 1027)
    myPBX.sip > ec2-34-226-36-35.compute-1.amazonaws.com.sip: SIP, length: 999
        SIP/2.0 404 Not Found
        Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bK3438.24c2c0a277d1fe3eef03b45aaf36f737.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK9414134
        Record-Route: <sip:34.226.36.35;lr>
        Call-ID: 75694be1-a936dd09-8dc5fe8@0.0.0.0
        From: <sip:ping@invalid>;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-4438dc77
        To: <[techprefix]@[my ap addr]>;tag=z9hG4bK3438.24c2c0a277d1fe3eef03b45aaf36f737.0
        CSeq: 1 OPTIONS
        Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary
, application/sdp, message/sipfrag;version=2.0
        Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
        Supported: 100rel, timer, replaces, norefersub
        Accept-Encoding: identity
        Accept-Language: en
        Server: Asterisk PBX 18.4.0
        Content-Length:  0

13:39:15.072475 IP (tos 0x0, ttl 52, id 32581, offset 0, flags [none], proto UDP (17), length 559)
    ec2-34-226-36-35.compute-1.amazonaws.com.sip > myPBX.sip: SIP, length: 531
        OPTIONS sip:[techprefix]@[my ap addr]:5060 SIP/2.0
        Max-Forwards: 20
        Record-Route: <sip:34.226.36.35;lr>
        Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bKa575.e458375d81faaa0f66b9e31d411530a3.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK8958333
        Route: <sip:34.226.36.35:5060;lr;received=sip:[my ip addr]:5060>
        From: sip:ping@invalid;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-e2badc77
        To: sip:[techprefix]@[my ip addr]:5060
        Call-ID: 75694be1-48b8dd09-01d5fe8@0.0.0.0
        CSeq: 1 OPTIONS
        Content-Length: 0
        Max-Forward: 10
        
13:39:15.073063 IP (tos 0x0, ttl 64, id 23266, offset 0, flags [DF], proto UDP (17), length 1027)
    myPBX.sip > ec2-34-226-36-35.compute-1.amazonaws.com.sip: SIP, length: 999
        SIP/2.0 404 Not Found
        Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bKa575.e458375d81faaa0f66b9e31d411530a3.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK8958333
        Record-Route: <sip:34.226.36.35;lr>
        Call-ID: 75694be1-48b8dd09-01d5fe8@0.0.0.0
        From: <sip:ping@invalid>;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-e2badc77
        To: <sip:[techprefix]@[my ip addr]>;tag=z9hG4bKa575.e458375d81faaa0f66b9e31d411530a3.0
        CSeq: 1 OPTIONS
        Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary
, application/sdp, message/sipfrag;version=2.0
        Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
        Supported: 100rel, timer, replaces, norefersub
        Accept-Encoding: identity
        Accept-Language: en
        Server: Asterisk PBX 18.4.0
        Content-Length:  0
        
13:40:10.350384 IP (tos 0x0, ttl 52, id 45055, offset 0, flags [none], proto UDP (17), length 559)
    ec2-34-226-36-35.compute-1.amazonaws.com.sip > myPBX.sip: SIP, length: 531
        OPTIONS sip:[techprefix]@[my ip addr]:5060 SIP/2.0
        Max-Forwards: 20
        Record-Route: <sip:34.226.36.35;lr>
        Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bK9f07.9d03ede3aff6720da0885b1f2bfe8490.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK3773069
        Route: <sip:34.226.36.35:5060;lr;received=sip:[my ip addr]:5060>
        From: sip:ping@invalid;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-f03ddc77
        To: sip:[techprefix]@[my ip addr]:5060
        Call-ID: 75694be1-563bdd09-74d5fe8@0.0.0.0
        CSeq: 1 OPTIONS
        Content-Length: 0
        Max-Forward: 10
        
13:40:10.350983 IP (tos 0x0, ttl 64, id 60461, offset 0, flags [DF], proto UDP (17), length 1027)
    myPBX.sip > ec2-34-226-36-35.compute-1.amazonaws.com.sip: SIP, length: 999
        SIP/2.0 404 Not Found
        Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bK9f07.9d03ede3aff6720da0885b1f2bfe8490.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK3773069
        Record-Route: <sip:34.226.36.35;lr>
        Call-ID: 75694be1-563bdd09-74d5fe8@0.0.0.0
        From: <sip:ping@invalid>;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-f03ddc77
        To: <sip:[techprefix]@[my ip addr]>;tag=z9hG4bK9f07.9d03ede3aff6720da0885b1f2bfe8490.0
        CSeq: 1 OPTIONS
        Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
        Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
        Supported: 100rel, timer, replaces, norefersub
        Accept-Encoding: identity
        Accept-Language: en
        Server: Asterisk PBX 18.4.0
        Content-Length:  0
        
13:41:05.643466 IP (tos 0x0, ttl 52, id 54737, offset 0, flags [none], proto UDP (17), length 559)
    ec2-34-226-36-35.compute-1.amazonaws.com.sip > myPBX.sip: SIP, length: 531
        OPTIONS sip:[techprefix]@[my ip addr]:5060 SIP/2.0
        Max-Forwards: 20
        Record-Route: <sip:34.226.36.35;lr>
        Via: SIP/2.0/UDP 34.226.36.35:5060;branch=z9hG4bK08a7.c282f095080e3c7cb326601ef4a52204.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK9368562
        Route: <sip:34.226.36.35:5060;lr;received=sip:[my ip addr]:5060>
        From: sip:ping@invalid;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-2fafdc77
        To: sip:[techprefix]@[my ip addr]:5060
        Call-ID: 75694be1-84bddd09-e7d5fe8@0.0.0.0
        CSeq: 1 OPTIONS
        Content-Length: 0
        Max-Forward: 10
        
13:41:05.644052 IP (tos 0x0, ttl 64, id 9841, offset 0, flags [DF], proto UDP (17), length 1027)
    myPBX.sip > ec2-34-226-36-35.compute-1.amazonaws.com.sip: SIP, length: 999
        SIP/2.0 404 Not Found
        Via: SIP/2.0/UDP 34.226.36.35:5060;rport=5060;received=34.226.36.35;branch=z9hG4bK08a7.c282f095080e3c7cb326601ef4a52204.0
        Via: SIP/2.0/UDP 52.40.141.109:5060;branch=z9hG4bK9368562
        Record-Route: <sip:34.226.36.35;lr>
        Call-ID: 75694be1-84bddd09-e7d5fe8@0.0.0.0
        From: <sip:ping@invalid>;tag=uloc-63881c7f-1e-f742ab26-0e1ec1f6-2fafdc77
        To: <sip:[techprefix]@[my ip addr]>;tag=z9hG4bK08a7.c282f095080e3c7cb326601ef4a52204.0
        CSeq: 1 OPTIONS
        Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
        Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UP

Note: [my ip addr] (in the above code) references (and matches) “my pub ip addr” from config in the OP. It is the external, and public, IP address.

If this doesn’t help could someone point me to the log collection instructions? I have searched but without success.

Thank you very much!

7

Extra tcpdump data showing:

  1. received msgs from flowroute (but no response)
  2. received msg from my phone with response

I’m just digging for more information–I hope this isn’t too much.

14:59:22.980464 IP myPBX.57139 > _gateway.domain: 7651+ [1au] PTR? 33.36.226.34.in-addr.arpa. (54)
14:59:22.993029 IP _gateway.domain > myPBX.57139: 7651 1/0/1 PTR ec2-34-226-36-33.compute-1.amazonaws.com. (108)
14:59:23.319267 IP ec2-34-226-36-33.compute-1.amazonaws.com.sip > myPBX.sip: SIP: INVITE sip:[my phone number]@[my ip addr]5060 SIP/2.0

14:59:23.649788 IP ec2-34-226-36-33.compute-1.amazonaws.com.sip > myPBX.sip: SIP: OPTIONS sip:[techprefix]@[my ip addr]:5060 SIP/2.0

14:59:24.046705 IP ec2-34-226-36-33.compute-1.amazonaws.com.sip > myPBX.sip: SIP: OPTIONS sip:[techprefix]@[my ip addr]:5060 SIP/2.0
14:59:24.109504 IP ec2-34-226-36-33.compute-1.amazonaws.com.sip > myPBX.sip: SIP: INVITE sip:[my phone number]@[my ip addr]:5060 SIP/2.0

14:59:24.859177 IP ec2-34-226-36-33.compute-1.amazonaws.com.sip > myPBX.sip: SIP: OPTIONS sip:[techprefix]@[my ip addr]:5060 SIP/2.0


14:59:30.426117 IP myphone.55770 > myPBX.sip: SIP

14:59:30.468507 IP myPBX.36060 > _gateway.domain: 51844+ [1au] PTR? [my phone private net ip addr].in-addr.arpa. (53)
14:59:30.469443 IP _gateway.domain > myPBX.36060: 51844* 1/0/1 PTR myphone. (74)

Edit: Update: I cleared /var/log/asterisk and restarted the server. The server started out normally but then something stopped working. I did pjsip set logger on (and debug, verbose = 5). Here is the entire log:

[Dec 11 16:48:18] Asterisk 18.4.0 built by mockbuild @ buildvm-x86-15.iad2.fedoraproject.org on a x86_64 running Linux on 2022-06-27 10:23:26 UTC
[Dec 11 16:48:18] NOTICE[89299] loader.c: 310 modules will be loaded.
[Dec 11 16:48:18] NOTICE[89299] cdr.c: CDR simple logging enabled.
[Dec 11 16:48:18] WARNING[89299] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Dec 11 16:48:18] WARNING[89299] res_phoneprov.c: Unable to find a valid server address or name.
[Dec 11 16:48:18] NOTICE[89299] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Dec 11 16:48:19] WARNING[89299] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23.
[Dec 11 16:48:19] WARNING[89299] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Dec 11 16:48:19] WARNING[89299] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35.
[Dec 11 16:48:19] WARNING[89299] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Dec 11 16:48:19] WARNING[89299] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Dec 11 16:48:19] ERROR[89299] ari/config.c: No configured users for ARI
[Dec 11 16:48:19] NOTICE[89299] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Dec 11 16:48:19] NOTICE[89299] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Dec 11 16:48:19] ERROR[89299] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
[Dec 11 16:48:19] WARNING[89299] loader.c: Some non-required modules failed to load.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'res_monitor' has been loaded but may be removed in a future release. Its replacement is 'app_mixmonitor'.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'res_adsi' has been loaded but may be removed in a future release.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'app_getcpeid' has been loaded but may be removed in a future release.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'app_image' has been loaded but may be removed in a future release.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'app_macro' has been loaded but may be removed in a future release. Its replacement is 'app_stack (GoSub)'.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'app_nbscat' has been loaded but may be removed in a future release.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'app_url' has been loaded but may be removed in a future release.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'app_dahdiras' has been loaded but may be removed in a future release.
[Dec 11 16:48:19] WARNING[89299] loader.c: Module 'app_adsiprog' has been loaded but may be removed in a future release.
[Dec 11 16:48:19] ERROR[89299] loader.c: Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Dec 11 16:48:19] ERROR[89299] loader.c: res_timing_dahdi declined to load.
[Dec 11 16:48:19] ERROR[89299] loader.c: cdr_syslog declined to load.
9

After more research I think I have improved on log collection. Here is a log from server-startup to failure.

I’m hping I have, now, correctly collected the necessary log detail and this post will get me closer to a solution.

Can someone please help? Thank you very much, in adavnce.

Bad log deleted.

10

Enable and use the full log.

Turn the debug level down.

Make sure verbosity is at least 3.

Make sure that “pjsip set logger on” has been enabled.

11

Full log enabled.
debug 5
verbosity 5
pjsip set logger on

I’ll post as soon as I get a fail.

Thank you very much!

12

It started failing surprisingly fast this time.

Here’s the log. I hope I got everything set-up correctly for logging and yes, as stated, I followed your instructions.

full-to-fail-cln.txt (207.3 KB)

13

In hopes of providing better information, here is a new log (with failure) and the associated tcpdump showing flowroute’s messages to me and (I think) no response from Asterisk.

full-to-fail-231219-1025EST-cln.txt (2.3 MB)
tcpdump-231219-1025EST.txt (6.0 KB)

Does this help?
Did I get the logging correct (full with debug 5, verbose 5, pjsip set logger on)?

14 
[Dec 18 22:26:46] VERBOSE[108981][C-00000001] file.c: <PJSIP/flowroute-00000000> Playing 'custom/no-solicitors.slin' (language 'en')
[Dec 18 22:26:48] VERBOSE[108936] res_pjsip_logger.c: <--- Received SIP request (599 bytes) from UDP:34.226.36.32:5060 --->
BYE sip:my-pub-ip-addr:5060 SIP/2.0
Record-Route: <sip:34.226.36.32;lr>
Max-Forwards: 67
Record-Route: <sip:34.211.73.216;lr>
From: "MIRMAN LISA" <sip:+1myDID@fl.gg>;tag=gK081d8125
To: <sip:+1myflowroute-in-DID@fl.gg>;tag=e8548d42-0bc9-4da0-b7a5-26282cc9a4d9
Via: SIP/2.0/UDP 34.226.36.32:5060;branch=z9hG4bK84d9.ccf3d4d51ac72e8f868cca6092b2ae21.0
Via: SIP/2.0/UDP 34.211.73.216:5060;branch=z9hG4bK84d9.73328b901d2e6ce4d8ca0739609e0987.0
Via: SIP/2.0/UDP 207.223.78.224:5060;branch=z9hG4bK08B8f2a4259c528837c
Call-ID: 476613197_32483484@207.223.78.224
CSeq: 81624 BYE
Content-Length:   0

The caller ended the call whilst you were playing the initial message, which would be typical of answering machine detection, although it is also possible that they were not receiving RTP.

It was a normal call clearing and they did not provide any additional information about the reason.

It might be worth sending Ringing, and giving them a few seconds, in case they have a strange requirement that all calls have ringing.

I also note that they have not sent any media to you.

[Dec 18 22:26:43] DEBUG[108981][C-00000001] channel.c: Didn't receive a media frame from PJSIP/flowroute-00000000 within 500 ms of answering. Continuing anyway

That means you can’t learn their real media address, so if that was not correct in the SDP, the announcement won’t reach them.

15

Hi,

The caller ended the call

That was me initiating a test call. When I heard the message I hung-up.

I am now waiting longer on my test-calls as you will see. Please note I did accidentally hang-up early one one or two.

Would I be correct in using “ringing” from extensions.conf
within my dialplan, before the message plays, in the form:

exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
exten => s,n,Wait,2

If the above is correct I’ll implement that to test.

I believe I resolved this and it does not appear in the latest full log.

I’m attaching the log with hopes someone can spot why my PBX stops responding to flowroute.

Thank you very much for your help!

full-to-fail-231220-1816EST-cln.txt (712.1 KB)

16

@david551 Update:

Following your recommendation, I have changed my extensions.conf (given the possibility of flworoute requiring ringing):

  exten => mynumber,1,Ringing
   same => n,Wait(3)
   same => n,Answer()

I’m now logging, and waiting, to see what happens.

17

OK, with your suggestions in play, as stated above, my Asterisk PBX just failed.

Here’s the log. My fingers are crossed it shows something.

full-to-fail-231222-1131EST-cln.txt (456.7 KB)

18

Again the caller hung up normally during the announcement, and, again, there is a lack of incoming media.

What is 34.226.36.33?

19

I noticed the failure right before I posted–and I just noticed these posts do not appear to be time-stamped.

The test-calls were me verifying answer. Is there something else I could do to avoid the issue of to-early dc? Or is there a better way I can test? The only way I know there is an issue is (easy-check) when I get a “number disconnected” message from flowroute or (more challenging) I watch tcpdump for my server to stop responding to inbound flowroute messages.

34.226.36.33 is part of flowroutes service:

34.210.91.112/28  # US-West-OR(.sip.flowroute.com) 34.210.91.112-126
34.226.36.32/28   # US-East-VA 34.226.36.32-46
16.163.86.112/30  # AP-East-HK
3.0.5.12/30       # AP-SouthEast-SIN
3.8.37.20/30      # EU-West-LDN
3.71.103.56/30    # EU-Central-FRA
18.228.70.48/30   # SA-East-SP

Thank you!

20

I’m getting confused. If these are test calls and not the failing calls, which part of the log do I need to look at to see a failing call. Until I understand how the call is failing, I can’t suggest a solution.

21

I’m very sorry–let me see if I can do a better job. I’ll treat the like a bug report.

Problem: Asterisk PBX stops responding to flowroute.
Additional symptom: dialing DID results in flowroute playing “number not in service” message.

Steps to reproduce:

  1. Start server
  2. Call DID, verify Asterisk responds to flowroute and answers, properly.
  3. Wait unspecificed (possibly random) time.
  4. Call DID, hear “number not in service message” and see inbound sip messages from flowroute without expected response from Asterisk PBX.

Restarting server typically starts back at step-1.

I am attaching a fresh full log and tcpdump. Looking at the full log (at the end) and comparing to the tcpdump, you should see messages from flowroute to which Asterisk does not respond (specifically towards the end–I stopped as soon as I found it failing to collect the log and tcpdump output.

I sincerely hope this clarifies things–I’m worried about making things as straight-forward and easy for you as possible.

Thank you very much for your help!!!

full-to-fail-231222-1606EST-cln.txt (375.0 KB)
watchtcpip231222-13-29-08-cln.txt (405.5 KB)