sipXcom 25.01.0-rc-002 Proxy/Registrar Issue

[SipXcom Newbie]

Good morning,

- Fresh bare metal install.
- Switched every telephone service on
- created a single "user"
- rebooted


# service sipxecs status
sipxconfig ( pid 30463 ) is running and listening on port
sipxbridge ( pid 5276 ) is running and listening on port
sipxcallback is stopped
sipxcdr ( pid 6153 ) is running
freeswitch ( pid 8305 ) is running and listening on port
sipxivr ( pid 4000 ) is running and listening on port
sipxlogwatcher ( pid 2933 ) is running
sipxpublisher ( pid 8830 ) is running and listening on port
sipxrelay ( pid 3847 ) is running and listening on port
sipxsqa ( pid 9363 ) is running and listening on port
sipxpage ( pid 2643 ) is running and listening on port
sipxproxy ( pid 28064 ) is running and listening on port
sipxregistrar ( pid 8964 ) is running and listening on port
sipxrest ( pid 3099 ) is running and listening on port
sipxrls ( pid 9505 ) is running and listening on port
sipxsaa ( pid 9278 ) is running and listening on port
sipxsupervisor ( pid 1082 ) is running
sipxrecording ( pid 2840 ) is running and listening on port


# netstat -a | grep :5060
#


The proxy log contains:

13:KERNEL:ERR:fqdn::sipxproxy:"OsConnectionSocket(ip:9090): call to connect() failed with error: 115 'Operation now in progress'"
14:HTTP:ERR:fqdn::sipxproxy:"HttpMessage::get[4] socket to ip:9090 not connected, retry 1 after 20ms"
15:KERNEL:ERR:fqdn::sipxproxy:"OsConnectionSocket(ip:9090): call to connect() failed with error: 115 'Operation now in progress'"
16:HTTP:ERR:fqdn::sipxproxy:"HttpMessage::get[4] socket to ip:9090 not connected, retry 2 after 40ms"
17:HTTP:ERR:fqdn::sipxproxy:"HttpMessage::get[4] socket connection to ip:9090 failed, give up...\n"
18:XMLRPC:ERR:fqdn::sipxproxy:"XmlRpcRequest::execute http connection failed"
19:NAT:CRIT:fqdn::sipxproxy:"MediaRelay::executeAndValudateSymmitronRequest failed to execute() request: -6 : Connection Failed"
20:NAT:CRIT:fqdn::sipxproxy:"MediaRelay::initialize() sign in failed: -6:'Connection Failed'"
21:SIP:NOTICE:fqdn::sipxproxy:"E911: Emergency Line Identifier successfully loaded /etc/sipxpbx/authrules.xml"
22:SIP:NOTICE:fqdn::sipxproxy:"SipBidirectionalProcessorPlugin::905_gatewaydest CREATED"
23:SIP:NOTICE:fqdn::sipxproxy:"SipBidirectionalProcessorPlugin::announceAssociatedSipUserAgent Name=905_gatewaydest Address=ip:5060 ASSIGNED"

[Support]

Thanks for contributing status. Much better!

The 'is running and listening on port’ statuses are strong indications that the services are OK, live and responsive. Same if it just says ’is running’.

The logs may still kick up some errors during startup until all components and subsystems are ready.

We will check why the sipxcallback service is showing as stopped. It is not a service we normally enable here, so something may have slipped through the cracks. We are seeing the same stopped status after enabling the service here even if the actual callback service appears to be running.

As you start adding users make sure you update to the latest 25.01 update 6 that was just published yesterday. We fixed an important issue there with the search index refresh and associated config replication after user additions and deletions.

Now that you are ready to start adding users and trunks, and test some actual SIP calls, don’t be shy asking for help.

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[SipXcom Newbie]

Dear OnRelay Support,

Thank you, as always, for your prompt and helpful support.

Regarding enabled services, as the point is evaluation rather than
production, testing commenced with everything enabled.

Regarding adding users and trunks, I take it that I should see a
listener on 5060 directly after enabling all services in the UI
*without need of creating a user or device*?

Regarding creating a user, I reported that aliases could not be
deleted and trust that was fixed...

One of the reasons for trying to evaluate sipXcom was to understand
the ways it supports a m:n relationship between users and devices.

"Asterisk doesn’t really support the concept of Users - everything
is related to extensions. Users are a construct built as a bag on
the side of Asterisk."
-- https://community.freepbx.org/t/users-extensions-devices/66154

Every switch takes its own approach and having evaluated FreePBX it
wasn't found suitable.

On sipXcom, under User ID on the Create User UI screen, it says:
"The User ID is displayed by the phone and it is therefore recommended
to use the internal extension as the User ID."

This appears to imply a 1:1 between device and user.

Is there something that explains the model(s) sipXcom supports?
There doesn't appear to be a "Find Me" service, but that may be
because sipXcom takes a different approach.

Regarding nothing listening on 5060, if updating doesn't help, I'll
cold metal reinstall [again].


SipXcom Newbie

----- Original Message
Date: Fri, 19 Jun 2026 11:51:49 +0200
From: Support <sup...@onrelay.net>
To: SipXcom Newbie <sipxcom-text...@tel.co.uk>
Cc: Sipxco...@googlegroups.com
Subject: Re: [sipxcom-users] sipXcom 25.01.0-rc-002 Proxy/Registrar Issue

[SipXcom Newbie]

Good afternoon,

Upgraded, rebooted and waited half an hour.

# service sipxecs status
sipxconfig ( pid 8011 ) is running and listening on port
sipxbridge ( pid 4544 ) is running and listening on port
sipxcallback is stopped
sipxcdr ( pid 1150 ) is running
freeswitch ( pid 8147 ) is running and listening on port
sipxivr ( pid 3839 ) is running and listening on port
sipxlogwatcher ( pid 2818 ) is running
sipxpublisher ( pid 8665 ) is running and listening on port
sipxrelay ( pid 3825 ) is running and listening on port
sipxsqa ( pid 3095 ) is running and listening on port
sipxpage ( pid 3088 ) is running and listening on port
sipxproxy ( pid 3049 ) is running and listening on port
sipxregistrar ( pid 8793 ) is running and listening on port
sipxrest ( pid 3091 ) is running and listening on port
sipxrls ( pid 8950 ) is running and listening on port
sipxsaa ( pid 9049 ) is running and listening on port
sipxsupervisor ( pid 1089 ) is running
sipxrecording ( pid 3090 ) is running and listening on port

The proxy-alarms log contains:

19:NAT:CRIT:fqdn:sipxproxy:"MediaRelay::executeAndValudateSymmitronRequest failed to execute() request: -6 : Connection Failed"
20:NAT:CRIT:fqdn:sipxproxy:"MediaRelay::initialize() sign in failed: -6:'Connection Failed'"

7:NET:EMERG:fqdn:sipxproxy:"Unable to bind on tls port 5061 (ok=0)"
9:NET:EMERG:fqdn:sipxproxy:"Unable to bind on tcp port 5060 (ok=0)"

Nothing else is using them:

# netstat -a | grep :5060
#

The proxy-alarms log continues:

18:SIP:CRIT:fqdn::sipxproxy:"ALARM_DNS_LOOKUP_FAILED DNS lookup failed for 'rr.fqdn'. No valid 'SRV' records found"

Yet:
- UI DNS Advisor reports "DNS Configuation is valid"
- On the box, # host -t SRV _sip._udp.fqdn returns:
_sip._udp.fqdn has SRV record 0 5 5060 fqdn.
for example.

The proxy log has otherwise identical entries as reported earlier.

[Support]

Thanks!

Comments below.

> Regarding adding users and trunks, I take it that I should see a
> listener on 5060 directly after enabling all services in the UI
> *without need of creating a user or device*?

OR: Yes, the sipxproxy service listens for SIP extensions on 5060 and 5061, whereas sipxbridge listens for SIP trunks on 5080 and 5081.

Will respond to this separately as per your second post with logs.

> Regarding creating a user, I reported that aliases could not be
> deleted and trust that was fixed...

OR: We must have missed that, and it is still indeed a fault. Will be fixed next update. If severe / blocking you may recreate the user to workaround.

> One of the reasons for trying to evaluate sipXcom was to understand
> the ways it supports a m:n relationship between users and devices.
>
> "Asterisk doesn’t really support the concept of Users - everything
> is related to extensions. Users are a construct built as a bag on
> the side of Asterisk."
> -- https://community.freepbx.org/t/users-extensions-devices/66154
>
> Every switch takes its own approach and having evaluated FreePBX it
> wasn't found suitable.
>
> On sipXcom, under User ID on the Create User UI screen, it says:
> "The User ID is displayed by the phone and it is therefore recommended
> to use the internal extension as the User ID."
>
> This appears to imply a 1:1 between device and user.

OR: It is common to use a short number / extension number as a unique system user ID.

Extensions or ‘Phones' in sipXcom terminology have a many-to-many relationship with Users, meaning a User can be represented on many Phones and a Phone can have many Users. A User can also have no Phones, e.g. own just a voice mail box.

There is also a presence (RLS) service that supports in-call status for users across phones.

It is perhaps also worth mentioning we at OnRelay also extend this User ID to mobile / cellular users in our MBX service, which e.g. also supports parallel rings between cell phones and SIP phones.

To see the supported Phone models, go to the Devices - Phones panel and select the Add Phone dropdown there.

We are just about to share a 27.01 roadmap where a WebRTC client will be an important addition to sipXcom as well.

> Is there something that explains the model(s) sipXcom supports?
> There doesn't appear to be a "Find Me" service, but that may be
> because sipXcom takes a different approach.

OR: When we took over this project there was quite a lot of online knowledge and documentation lying around, and we took the straightforward avenue to import and continue the Wiki that Matt Keys from the previous sponsors had nearly completed. The wiki has a decent Features section, but not a comprehensive architecture overview.

There is however quite a lot of online documentation that is not maintained by us that has been written through the years, such as e.g. the Core Dial knowledge base https://docs.coredial.com.

> Regarding nothing listening on 5060, if updating doesn't help, I'll
> cold metal reinstall [again].

OR: It doesn’t look like there is anything wrong with your sipXcom installation now, so no need to reinstall. Will reply to your other post shortly.

[Support]

Hello,

— There are two issues that may be going on here:

1. The media relay may not have come up properly. It is quite resource demanding and there could be a resource exhaustion during startup, please check if it comes up OK if you run 'sipxecs service restart’ 

2. Unable to bind on 5060 and 5061 indeed could suggest something else in your environment is already occupying those ports. netstat -a doesn’t give any results here either, so try this command instead:

ss -tulpn | grep :5060

udp   UNCONN 0      0                           10.128.0.35:5060       0.0.0.0:*    users:(("sipXproxy",pid=152823,fd=11))                                                                                              

tcp   LISTEN 0      64                          10.128.0.35:5060       0.0.0.0:*    users:(("sipXproxy",pid=152823,fd=10))                                                                                              

An alternative explanation is similar to point 1. that your file descriptor resources or memory were exhausted during startup. Some of those recommended parameters may have changed with the Rocky Linux upgrade as various services and subsystems become ever more bloated.  Please e.g. ensure you have at least 2 CPUs and 8GB RAM.

— You can ignore the SRV record errors, only A records are necessary for a single server.

Again, your installation appears fine now, these issues seem environmental / HW related at a first glance and we can help bottoming them out.

Try to run sipxecs service restart after a reboot and check if it comes up OK then. Depending on outcome we will investigate logs next.

[Support]

Just adding 2c here, when sipxproxy says it is running and listening on port, this means it is bound to and listening on port 5060. 

So the errors from your logs seems to be earlier log statements from the startup sequence, perhaps before your server was entirely booted. 

I.e. doublecheck connectivity to port 5060 differently, looks like grep statement was incorrect, should be "netstat -a | grep 5060” not "netstat -a | grep :5060” (see below) since there can be some odd formatting of the host:port in the netstat result.  

Alternatively use "ss -tulpn | grep :5060” as suggested earlier.  From elsewhere try "openssl s_client -connect <IP or DN>:5061” as well to ensure nothing security wise is blocking the port.

# netstat -a | grep 5060

tcp        0      0 xxx.yyy.onrelay.n:15060 0.0.0.0:*               LISTEN     

udp        0      0 xxx.yyy.onrelay.n:15060 0.0.0.0:*                          

# netstat -a | grep :5060

(empty)

#

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[SipXcom Newbie]

Thanks for the helpful reply.


Listeners

| OR: Yes, the sipxproxy service listens for SIP extensions on 5060
| and 5061, whereas sipxbridge listens for SIP trunks on 5080 and 5081.

Thanks. Perhaps when the documentation is updated, those ports and
names of listening processes together with the UDP RDP ranges might be
helpful.


UI - Deletion of entries from Aliases

| OR: We must have missed that, and it is still indeed a fault. Will
| be fixed next update.

Thanks.


Identification of Users & Devices

> This appears to imply a 1:1 between device and user.

| OR: It is common to use a short number / extension number as a unique
| system user ID.

Agreed, but in evaluating various offerings, an implied "Extension
Number" model often points to a more PBX use case than desired here.

I wish to reduce the possibility of extension number guessing.

c.f. sendmail or postfix accepting mail for delivery that is addressed
u...@your-domain.com


Users & Devices Model

| Extensions or ‘Phones' in sipXcom terminology have a many-to-many
| relationship with Users, meaning a User can be represented on many
| Phones and a Phone can have many Users.

Great, so what's the intersection entity?

| It is perhaps also worth mentioning we at OnRelay also extend this
| User ID to mobile / cellular users in our MBX service, which e.g.
| also supports parallel rings between cell phones and SIP phones.

Given near universal smart phone ownership today, this is arguably now
more useful than "Follow Me". Done here too, but using a SIP client.

| To see the supported Phone models, go to the Devices - Phones panel
| and select the Add Phone dropdown there.

Seen that - great. Need here is less auto-provisioning (use custom
config) than directory ("Contacts" or Phone Book) maintenance.


Roadmap

| We are just about to share a 27.01 roadmap where a WebRTC client
| will be an important addition to sipXcom as well.

Great. Now, a WebRTC conference gateway would be a "killer app"...


Achitecture

| OR: When we took over this project there was quite a lot of online

| knowledge and documentation lying around, and we took the straight-

| forward avenue to import and continue the Wiki that Matt Keys from
| the previous sponsors had nearly completed. The wiki has a decent
| Features section, but not a comprehensive architecture overview.
|
| There is however quite a lot of online documentation that is not
| maintained by us that has been written through the years, such as
| e.g. the Core Dial knowledge base https://docs.coredial.com.

Good illustration of hooks that have successfully been implemented

Additional WiKis are useful but are no real substitute for a pukka
vendor manual with numbered sections and paragraphs that sets out
architecture, model, configuration etc., but hey, the additional
WiKis are useful so thanks!


Proxy/Registrar Issue

| OR: It doesn’t look like there is anything wrong with your sipXcom
| installation now, so no need to reinstall. Will reply to your other
| post shortly.

Thank you kindly.

Will read and follow before rushing to come back. ;-)

SipXcom Newbie

[Support]

Thank you.

Just some comments to comments ;)

• The ports for a SIP proxy defined in the SIP standard (RFC 3261) are 5060 and 5061, so for the sipxproxy service the port assignment are standardized, but for all the other services a port map would indeed be very useful. Noted.

• Alias issue has been fixed in upcoming update 7

• The many-to-many relationship between users and phones is maintained by the ‘Lines’ list for each Phone. Each Phone can have multiple lines (also sometimes referred to as Keys), each line representing a unique User.

• User ID can be anything. Early SIP standards had an expectation that regular phone numbers would eventually disappear, and sipXecs is originally designed that way. So it is entirely possible to use e.g. email addresses as user IDs and just configure internal / extension / short numbers as aliases, just as DIDs are added as aliases as well. Just beware nearly all real implementations uses the internal short number as user ID, and there is always and element of risk in departing from such commonplace configs.

• Also note sipXecs is tightly integrated with openfire, which allows XMPP user IDs as IM IDs on the format you indicate below. We have stepped up openfire version to the latest in 25.01, and implemented the corresponding integration and API changes, thereby also closing some ugly legacy security hoies, but admittedly this is not something we have had time to test extensively yet. We will next also take a close look at how openfire is dealing with WebRTC vs the freeswitch approach, since a lot is going on with Ignite and XMPP in this regards. XMPP is by no means a dead end.

• Note SIP clients and smartphones are not always a great match due to battery and OS background constraints, so a SIP client alone will not be able to receive smartphone calls unless a non-standard mechanism to handle push notifications for incoming call wake ups are added. None of this is trivial, since both Apple and Android change those mechanisms and permissions constantly.

• It is quite straightforward to add or update custom phone config files to the auto proviosioning mechanism.  The Phone Groups also allows you to do many system wide Phone settings from the web UI. 

• The existing sipXcom conference server will support WebRTC as well. Web RTC is just SIP over websockets, so from the sipXcom perspective a WebRTC connection is just a regular SIP user.

• As for the sipXproxy issue, this is not directly related, but we are working on a timing related problem for some weaker HW platforms that the sipxrelay service is not always ready when sipxproxy starts after a reboot. And this can again block NAT SIP registrations until sipxproxy is restarted (or the entire sipxecs for that matter). We are expecting to release update 7 shortly when this issue has been fixed.

[Michael Jackson]

Thanks for sharing experiences and advice. All this speaks to sipXcom together with its support being a strong candidate :)

SipXcom Newbie

------- Original Message ------
Date: Mon, 22 Jun 2026 15:24:34 +0200
From: Support <sup...@onrelay.net>
To: SipXcom Newbie <sipxcom-text...@tel.co.uk>
Cc: Sipxco...@googlegroups.com
Subject: Re: [sipxcom-users] sipXcom 25.01.0-rc-002 Proxy/Registrar Issue

[SipXcom Newbie]

OR said:

| - There are two issues that may be going on here:

|
| 1. The media relay may not have come up properly.
|

| 2. Unable to bind on 5060 and 5061 indeed could suggest something

| else ... so try this command instead:

|
| ss -tulpn | grep :5060
| udp UNCONN 0 0 10.128.0.35:5060 0.0.0.0:* users:(("sipXproxy",pid=152823,fd=11))
| tcp LISTEN 0 64 10.128.0.35:5060 0.0.0.0:* users:(("sipXproxy",pid=152823,fd=10))

My bad, I left-off the 'n'. The proxy was listening all along...

I should have reported the symptom originally: User not registering;
instead REGISTERs sent but the proxy returns 408 Request Timeouts.

SipXcom Newbie

[Support]

We are working on this issue at the moment, testing a fix as we speak, it happens after reboot on some HW platforms. The fix will be included in update 7.

The root cause seems to be a persistent http connection from the media relay service not being released properly if it fails first try, and subsequent retries on the closed persistent connections just time out. It is an old error in the C++ OSS core that may have caused some grief through the years. Safe fix.

In the meantime, just restarting the sipxrproxy service (service sipxproxy restart) from a terminal, alternatively all services (service sipxecs restart) recovers it.

[SipXcom Newbie]

Thanks for the quick response.

# service sipxproxy restart
Stopping sipXproxy: [ OK ]
Starting sipXproxy: [ OK ]

Unfortunately the proxy still returns 408.

Generally test with Retries=1 to avoid fail2ban banning, but the 408
is sent directly after the first try.

On the up side, feel less bad about not even getting far in months.

Thanks, happy to wait for the fix.

SipXcom Newbie

----- Original Message ------
Date: Mon, 22 Jun 2026 17:21:01 +0200
From: Support <sup...@onrelay.net>
To: SipXcom Newbie <sipxcom-text...@tel.co.uk>
Cc: Sipxco...@googlegroups.com
Subject: Re: [sipxcom-users] sipXcom 25.01.0-rc-002 Proxy/Registrar Issue

[Support]

Try 'service sipxecs restart' if restarting only sipxproxy is not enough.

[SipXcom Newbie]

Thanks.

Having disabled tries on the phone, it chunters through disabling and enbling but when finished and registration enabled on the one phone, unfortunately it still sent an immediate 408.

----- Original Message -----
Date: Mon, 22 Jun 2026 18:00:03 +0200
From: Support <sup...@onrelay.net>
To: SipXcom Newbie <sipxcom-text...@tel.co.uk>
Cc: Sipxco...@googlegroups.com
Subject: Re: [sipxcom-users] sipXcom 25.01.0-rc-002 Proxy/Registrar Issue

[OnRelay Support]

You can check in the sipXproxy.log if it is the same issue, you should see the media relay starting up OK there. 

Also make sure SIP ALG is disabled on your router if you are using the remote NAT function, alternatively register on TLS / 5061.

[SipXcom Newbie]

sipxproxy:"MediaRelay::initialize() sign in failed: -1:'Initialization in progress'"
[Full log below]

- Earlier tests done in the office - internal phones traversing single router
- That router, in use many years, is definitely is not doing SIP ALG
- This test done from from home
- Home router definitely has SIP ALG disabled
- Multi-line phones used in the office and at home for years

AS it's a known issue being addressed, I'll just wait. Thanks.

SipXcom Newbie

```
Full Log (edited)
2026-06-22T17:01:19.869541Z:1:MONGO_CLIENT:NOTICE:fqdn:sipxproxy:"[mongocxx] libmongoc logging callback enabled"
2026-06-22T17:01:19.869615Z:2:SIP:NOTICE:fqdn:sipxproxy:"sipxproxy initialized"
2026-06-22T17:01:19.869727Z:3:SIP:WARNING:fqdn:sipxproxy:"SipXproxymain::proxy large default expires value: 300 NOT RECOMMENDED"
2026-06-22T17:01:19.869874Z:4:SIP:NOTICE:fqdn:sipxproxy:"T1 Timer set by config. Value: 100 ms"
2026-06-22T17:01:19.869888Z:5:SIP:NOTICE:fqdn:sipxproxy:"Retransmit count set by config. Value: 4 times"
2026-06-22T17:01:19.870610Z:6:KERNEL:NOTICE:fqdn:sipxproxy:"OsTimer::TimerService STARTED."
2026-06-22T17:01:19.870870Z:7:SIP:NOTICE:fqdn:sipxproxy:"SIP Timer Values - mUnreliableTransportTimeoutMs : 100 mMaxResendTimeoutMs : 800 mTransactionStateTimeoutMs : 8000 mDefaultExpiresSeconds : 180 mDefaultSerialExpiresSeconds : 20"
2026-06-22T17:01:19.870945Z:8:SIP:WARNING:fqdn:sipxproxy:"SipUserAgent::setDefaultExpiresSeconds large expiresSeconds value: 300 NOT RECOMMENDED"
2026-06-22T17:01:19.870954Z:9:SIP:WARNING:fqdn:sipxproxy:"SipUserAgent::setInviteTransactionTimeoutSeconds large expiresSeconds value: 300 NOT RECOMMENDED"
2026-06-22T17:01:19.871024Z:10:SIP:NOTICE:fqdn:sipxproxy:"SipUserAgent::handleCancelQueue - STARTED"
2026-06-22T17:01:19.880171Z:11:SIP:NOTICE:fqdn:sipxproxy:"SipRouter::SipRouter Skipping IP address-based domain alias 'ip.add.re.ss'"
2026-06-22T17:01:19.880196Z:12:SIP:NOTICE:fqdn:sipxproxy:"BranchId::setSecret reset identifier key; previously generated branch ids will not be recognized as local."
2026-06-22T17:01:19.967067Z:13:NAT:CRIT:fqdn:sipxproxy:"MediaRelay::initialize() sign in failed: -1:'Initialization in progress'"
2026-06-22T17:01:28.057146Z:14:SIP:NOTICE:fqdn:sipxproxy:"E911: Emergency Line Identifier successfully loaded /etc/sipxpbx/authrules.xml"
2026-06-22T17:01:28.058197Z:15:SIP:NOTICE:fqdn:sipxproxy:"SipBidirectionalProcessorPlugin::905_gatewaydest CREATED"
2026-06-22T17:01:28.058334Z:16:SIP:NOTICE:fqdn:sipxproxy:"SipBidirectionalProcessorPlugin::announceAssociatedSipUserAgent Name=905_gatewaydest Address=ip.add.re.ss:5060 ASSIGNED"
2026-06-22T17:01:28.058351Z:17:SIP:NOTICE:fqdn:sipxproxy:"Transaction rejection is in effect when queue is at 75% capacity and transaction maximum count is 20000"
```

[OnRelay Support]

Yes, seems like the same issue that on your platform does not recover with a restart.

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[Support]

Note we are tracking this issue at: https://github.com/onrelay/sipxecs/issues/60