FOOD FOR THOUGHT No Matching endpoint found

dad311

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Just built a new PBX with ubuntu 1404 and Frepbx 12. I added an extension (sip_chan) and Im getting the following error message. What am I missing???? thx


[2014-11-18 22:42:08] NOTICE[3294]: res_pjsip/pjsip_distributor.c:250 log_unidentified_request: Request from '"House" <sip:[email protected]>' failed for '192.168.100.148:5061' (callid: [email protected]) - No matching endpoint found
 

xrobau

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You're slightly confused.

> I added an extension (sip_chan)

But...

> res_pjsip/pjsip_distributor.c:250 log_unidentified_request: Request from '"House"

By default, on a new installation, pjsip listens on 5060, and chan_sip listens on 5061. There's a big box telling you that when you create the extension.

Your options are:
  1. register to port 5061,
    or,
  2. change the port for chan_pjsip to 5062, and chan_sip to 5060.
 

IanL01

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You're slightly confused.

> I added an extension (sip_chan)

But...

> res_pjsip/pjsip_distributor.c:250 log_unidentified_request: Request from '"House"

By default, on a new installation, pjsip listens on 5060, and chan_sip listens on 5061. There's a big box telling you that when you create the extension.

Your options are:
  1. register to port 5061,
    or,
  2. change the port for chan_pjsip to 5062, and chan_sip to 5060.

This is similar if not the same as the problem I reported as "SPA3102 with Asterisk 12 SIP". I have tired

(a) setting the SPA3102 to port 5061, with FreePBX 12 set to the default ports, No Joy.

(b) changing the default port in FreePBX around and leaving the SPA defaulted to 5060, No Joy.

Local phones will register if I use chan_pjsip extensions.
Remote trunks will register using chan_sip trunks
Cannot get a local Trunk (SPA3102) to register using a chan_sip trunk (doing what I just described above, ensuring that both ends are using the same port).
Cannot get a local extension (SPA942) to register using a chan_sip extension (doing what I just described above, ensuring that both ends are using the same port).
 

dad311

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This is similar if not the same as the problem I reported as "SPA3102 with Asterisk 12 SIP". I have tired

(a) setting the SPA3102 to port 5061, with FreePBX 12 set to the default ports, No Joy.

(b) changing the default port in FreePBX around and leaving the SPA defaulted to 5060, No Joy.

Local phones will register if I use chan_pjsip extensions.
Remote trunks will register using chan_sip trunks
Cannot get a local Trunk (SPA3102) to register using a chan_sip trunk (doing what I just described above, ensuring that both ends are using the same port).
Cannot get a local extension (SPA942) to register using a chan_sip extension (doing what I just described above, ensuring that both ends are using the same port).

Yeah, I have something very screwy going on with my Grandstream Handytone box. If I use a soft SIP client, no issues.

I can register one port on my Grandstream, but not two ports the same time. I wonder if its because one IP, with two SIP clients using port 5060???
 

geopeterwc

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Yeah, I have something very screwy going on with my Grandstream Handytone box. I can register one port on my Grandstream, but not two ports the same time. I wonder if its because one IP, with two SIP clients using port 5060???
Configuring a second account on the Grandstream box requires that you assign a different SIP port for it. For example, port 1 is typically assigned SIP port 5060; the second account on the box can be configured to SIP port 5061 within the settings on the box for the second account. The definition for the second extension in FreePBX will default to port 5060. In my experience I have to re-access the extension definition in FreePBX to change the SIP port assignment from 5060 to 5061.

The resulting box configuration will use one IP and is configured for two SIP clients, 5060 for analog port 1, and 5061 for port 2. The two analog ports will operate independently as a result.

You can confirm successful configuration from FreePBX by selecting Reports | Asterisk Info | SIP Info. Look for the extension number in the "Sip Peers" portion of the resulting display. You'll see the "Host" IP assignment will be the same for both extension numbers, and the "Port" column will show the SIP Port assignments (5060 and 5061).

I have done this successfully with "manual" configuration of my Grandstream HT502 devices numerous times. The only issue that I have with the box is the delay before initiating a call unless the number dialed is terminated with "#". Otherwise, these boxes work well with two registered accounts ... and two different extension numbers.

/Pete./
 

dad311

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chan_pjsip listens on 5061
chan_sip listens on 5060

I don't remember seeing an option to change the pjsip to a different port. Maybe I need to use chan_sip and edit the extentions to reflect 5060 and 5061.
 

IanL01

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Summary Incrediblepbx 12 running on ubuntu.

Start from a clean ubuntu and re-installed Incrediblepbx12 by following Ward's instructions.

On the FreePBX Advanced Setting changed SIP Channel to only chan_sip.
Applied the config and rebooted.
Built a SIP Trunk and tried to connect to an SPA942 phone.
Ran the command line in SIP DEBUG On
Saw no messages.
Changed SIP channel to both, Applied the config and rebooted.
Now start getting the No matching endpoint found message.
Running out of ideas of what to try next.
 

Dave Gray

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Hmmm. Had a bit of an AH-HA moment.

Chasing this problem with some of my stuff (FreePBX 12 and Asterisk 12 on a Cloudatcost server.)
Found out, when you go from SIP to PJSIP, FreePBX mangles the entry for the device in pjsip-endpoint.conf
You need to drop and recreate the extension, not just convert it. (Also found out, my work connection blocks port 5061 but allows
5060, wonder if they expect it to be TLS?)
 

Sean Branam

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Was anyone able to solve this?
I'm seeing the same issue.
I'm testing a new install of
Incredible PBX 12.0.70
'Incredible PBX GUI'

I have a Grandstream GXP2200, which was working perfectly on
Incredible PBX 12.0.65
'VoIP Server'
The exact same settings in the gui have been applied for the phone and I keep seeing "no matching endpoint found"

Any ideas would be helpful.
 

Sean Branam

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I have not.
I reverted back to my old install for now. *sigh*
I plan on doing more testing next week.

I will certainly update this, if I find anything.
 

wa4zlw

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I have grandstreams too. Using older versions, but if you need to use different SOURCE ports on even boundaries. 5060, 5062, etc. usually each account will default to the next available port unless you change it.

Make sure that is setup properly.

leon
 

dhoppy

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Same problem here with a Grandstream GXP2200. I just upgraded to Asterisk 13.6.
 

Graham_UK

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As Dave Gray has noticed there are problems with FreePBX updating pjsip-endpoint.conf. I've found that if extensions are imported from csv file you need to go to each extension in turn and re-submit. This seems to create the missing entries and allow the extensions to register.
 

apauna

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Same problem with linksys PAP2 I tried everything and finally rolled back to Asterisk (Ver. 11.20.0) on Ubuntu using this article http://nerdvittles.com/?p=9713 I could not get the software SIP phone to work from this article http://nerdvittles.com/?p=14183 I tried three different version even FreePBX v13 itself no luck and after a week without a phone I give up. I would like to use Asterisk 13 and FreePBX 12 on Ubuntu as outlined by nerdvittles, but I have to have a working phone too. Let me know if I can help troubleshoot this further I can setup a VM and test things out and would be glad to talk further about this.
 

dbaum

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Welcome to the nightmare. I am disappointed in Ward, for whom I have great respect, that his article showed success only with softphone. Spent a day on the phone with tech support at Vitelity (for whom he is the reseller at the special prices) and they are baffled at Level 1 and Level 2 support. They want to charge to solve the problem @ $125 / hour.
 

wa4zlw

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i played with this yesterday. On the SL version of 13-12.2 I turned SIP on (as I am using pjsip). I was unable to make any connection using the old CHAN_SIP. I changed ports on the pbx and the phone to no avail.

SO I am back on pjsip with all devices working except for the soft keys/lights and certain media won't play on the older grandstream IP phones (see other posts I made)

Leon
 

billsimon

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What troubleshooting techniques are you guys using? Are you turning on debugs and grabbing tcpdumps or just looking at the asterisk logs? It seems clear to me that the asterisk logs are not telling the whole story.

By the way, 5061 is the standard setting for SIP using TLS. It is 100% wrong to use that port for unencrypted SIP.
 
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