Error with outbound call

[erdia]

Good afternoon, I have correctly configured a PBX with SIP extensions and trunks, and receive calls and everything, but when you call me out I get an error "404 Not Found / INVITE from xxx.xxx.xxx:5060" and will not let me call . Exit rules I set the Prepend I of 0 and is removed when dialing. Help! Thank you

 

[leejor]

If you can receive calls from outside, I have to assume that your trunk settings are correct and it is registered properly (it is a VoIP provider, correct?). It may be an issue with your outbound rules. If you could post the 3CX log of an outbound call attempt, it may provide some additional information to help resolve. Which provider are you using this with? Did you confirm that you are indeed sending numbers to them, in a correct format?

 

[erdia]

If you can receive calls from outside, I have to assume that your trunk settings are correct and it is registered properly (it is a VoIP provider, correct?). It may be an issue with your outbound rules. If you could post the 3CX log of an outbound call attempt, it may provide some additional information to help resolve. Which provider are you using this with? Did you confirm that you are indeed sending numbers to them, in a correct format?

If , incoming calls are received correctly and directed to the IVR . Nose that more try , I think I've properly configured outbound rule , saying that press 0 before and then delete it.

Thanks

 

[ucs1]

any chance you can provide more information.

IE: Your outbound rules, example of number dialed and any associated logs.

 

[erdia]

any chance you can provide more information.

IE: Your outbound rules, example of number dialed and any associated logs.

Yes, of course (192.168.1.1 is my SIP Server of ISP of Spain

"12-jul-2016 11:26:50.183 Leg L:7.1[Extn] is terminated: Cause: BYE from PBX
12-jul-2016 11:26:50.131 Leg L:7.4[Line:10000>>93XXXXXXX] is terminated: Cause: 404 Not Found/INVITE from 192.168.1.1:5060
12-jul-2016 11:26:50.128 [CM503020]: Call(C:7): Normal call termination. Call originator: Extn:100. Reason: Not found
12-jul-2016 11:26:50.127 L:7.1[Extn] failed to reach Line:10000>>93XXXXXXX, reason Not Found
12-jul-2016 11:26:50.127 Call to T:Line:10000>>93XXXXXXX@[Dev:sip:*[email protected]:5060] from L:7.1[Extn] failed, cause: Cause: 404 Not Found/INVITE from 192.168.1.1:5060
12-jul-2016 11:26:50.126 [CM503003]: Call(C:7): Call to <sip:[email protected]:5060> has failed; Cause: 404 Not Found/INVITE from 192.168.1.1:5060"


But, if I register directly *[email protected] in my sofpthone any problem, and i the PBX has problem. Problem is in the PBX


And the rules of outbound calls
Attachment_1


And I don't know if this is relevant
Attachment_2


Thank you!!!

 

[leejor]

What are you using for the trunk (10000/livebox) ? It doesn't appear to be a VoIP provider as suggested by your initial post, and I questioned you about, but received no answer.

If you have some sort of gateway, it may be that the port you have datafilled in the 3CX trunk settings (for that trunk) is not the same as set in the device. That may be why, if the IP is correct, 3CX cannot find it.

Your outbound rules also show you prepending the number with +34. The + sign is generally used when a system knows to replace it with the local International dialling prefix, not when the final digits are sent to a device. So you might want to try removing it as whatever device the digits go to, may not understand it.

I'm only seeing one IP in the log (192.168.1.1), which I assume is the 3CX server PC. What is the IP of the gateway?

 

[erdia]

What are you using for the trunk (10000/livebox) ? It doesn't appear to be a VoIP provider as suggested by your initial post, and I questioned you about, but received no answer.

If you have some sort of gateway, it may be that the port you have datafilled in the 3CX trunk settings (for that trunk) is not the same as set in the device. That may be why, if the IP is correct, 3CX cannot find it.

Your outbound rules also show you prepending the number with +34. The + sign is generally used when a system knows to replace it with the local International dialling prefix, not when the final digits are sent to a device. So you might want to try removing it as whatever device the digits go to, may not understand it.

I'm only seeing one IP in the log (192.168.1.1), which I assume is the 3CX server PC. What is the IP of the gateway?

Thank you for your reply, I mean, in Spain there is an ISP that provides a router with voip and this router has the ability to be a proxy Sip, because the isp does not provide the SIP data directly, you have to use the SIP router, and as the router is 192.168 .1.1 as the SIP address that is. The gateway also because the router is 192.168.1.1 manages everything. And as I said, if I sign up directly with a softphone works perfectly

 

[leejor]

It sounds as if you are attempting to use a device/configuration that may , or may not be, supported. While I'm not saying that it won't work, it is not a set-up that I'm familiar with. If a SIP phone is able to register to it, then, in theory, a trunk from 3CX should be able to as well, in the same manner as a PSTN gateway. The 404 message may be because the device does not know what to do with the + that you are sending it. Remove that and prepend the digits (34) with any numbers required. Don't assume that it can "deal" with the +.

 

[Sopock]

12-jul-2016 11:26:50.126 [CM503003]: Call(C:7): Call to <sip:[email protected]:5060> has failed; Cause: 404 Not Found/INVITE from 192.168.1.1:5060"

maybe softphone is using server name like sip.orange.es :?:
I would also try on newer install of 3CX to call some toll free numbers.

 

[erdia]

12-jul-2016 11:26:50.126 [CM503003]: Call(C:7): Call to <sip:[email protected]:5060> has failed; Cause: 404 Not Found/INVITE from 192.168.1.1:5060"

maybe softphone is using server name like sip.orange.es :?:
I would also try on newer install of 3CX to call some toll free numbers.

I was intent and fail Any more reply? I'm nervous for this problem

 

[leejor]

maybe softphone is using server name like sip.orange.es

You aren't using a VoIP provider directly (as I understand it), you are using a router with some sort of a gateway built in, so the use of URLs is probably not an issue.. What IP address is your 3CX server using? Did you give it a fixed IP on the network? I'm assuming that the "box" acting as a Gateway, is also the router/DHCP server? Did you do any port forwarding on the router portion that may be interfering with where port 5060 is routing to?

You may need to provide more detail about how you have you network set up. is the router and gateway using the same IP (192.168.1.1) ?

Have you done any searches on the net using the make/model of the device you are using , along with "3CX", to see if anyone else has has success in using it? It may very well be some simple setting that is preventing it from working properly, but there may be others out there (in Spain), that have tried this before. They may have posted their results if you do some searches.

I just did a search for "Livebox" and the device that came up would seen to indicate that it has an RJ11 socket for an analogue phone. This would mean that it is an ATA. Are you trying to bypass the built in ATA and register directly with the provider?

 

[erdia]

maybe softphone is using server name like sip.orange.es

You aren't using a VoIP provider directly (as I understand it), you are using a router with some sort of a gateway built in, so the use of URLs is probably not an issue.. What IP address is your 3CX server using? Did you give it a fixed IP on the network? I'm assuming that the "box" acting as a Gateway, is also the router/DHCP server? Did you do any port forwarding on the router portion that may be interfering with where port 5060 is routing to?

You may need to provide more detail about how you have you network set up. is the router and gateway using the same IP (192.168.1.1) ?

Have you done any searches on the net using the make/model of the device you are using , along with "3CX", to see if anyone else has has success in using it? It may very well be some simple setting that is preventing it from working properly, but there may be others out there (in Spain), that have tried this before. They may have posted their results if you do some searches.

I just did a search for "Livebox" and the device that came up would seen to indicate that it has an RJ11 socket for an analogue phone. This would mean that it is an ATA. Are you trying to bypass the built in ATA and register directly with the provider?

I mean, because it's a little confusing. Orange (my ISP) works with VoIP telephony, but does not provide the SIP data because they say they work by connecting an analog phone to the router with RJ11 (I ATA built not configurable) what they want is to look like an analog phone, that is the intention Orange, why not provide SIP data. So because of this, the router includes a SIP server that functions as a PBX but the only thing that makes this PBX is managing the SIP data, then to configure phone ip pbx ip with orange, it must be done through the gateway (192.168.1.1) If you seek online Orange Livebox 2.1 softphone you find information about this. I've also changed the ports of the PBX because router when VoIP lets not map the port 5060. Then change it to port 6500 and redirect it to my PC. (192.168.1.103). The SIP address, also acts as a gateway / DHCP / modem / router / wifi. It is all in. Deputy captures the gateway for you to observe and thus solve my problem. Thank you!!

PS: The zoiper not register me the PBX do not know why. Change the port to 6500 in the zoiper too.
Attachment_1
Attachment_2

 

[lneblett]

Out of curiosity, why is there a "+" in the prepend for the outbound rules and in the e164 section? I also question why there are failover outbound routes, but all using the same name?

I am a little confused as you stated that you changed the SIP port to 6500; yet the log shows that 5060 is still in use.

 

[erdia]

Out of curiosity, why is there a "+" in the prepend for the outbound rules and in the e164 section? I also question why there are failover outbound routes, but all using the same name?

I am a little confused as you stated that you changed the SIP port to 6500; yet the log shows that 5060 is still in use.

I think the problem is in the +34 because it was not set. The port 5060 is the SIP proxy that connects the PBX to the router. And 6500 is the port to connect the extensions. What a mess! Probare to install it on another pc see what happens. In the meantime there if anyone knows. Thank you!

 

[leejor]

Well, as I thought, you are trying to make something work that was never interned by your provider, to work.
They have provided you with a box with a built in ATA, and don't give you the settings to allow you to set-up a "bring your own device" to place calls. It sounds as if they have blocked the forwarding (re-use) of port 5060 as they intend on the built in ATA to use that. This is not unique to Orange, or a device such as the Livebox, but to honest, I haven't actually heard of someone making this work (using a similar device), with 3CX. Maybe you''l be the first, but I don't give it much hope as the provider has probably "engineered" it to not work the way you want it to. They may even have a way of detecting that you have made modifications, trying to "get around" the ATA, which may be in violation of the terms of service.

In most cases, where a provider supplies a box, with FXS ports (dialtone), users will attach a gateway, then register that gateway as a trunk on the PBX. This is in the same manner, as would be done, with any PSTN line.

 

[erdia]

Well, as I thought, you are trying to make something work that was never interned by your provider, to work.
They have provided you with a box with a built in ATA, and don't give you the settings to allow you to set-up a "bring your own device" to place calls. It sounds as if they have blocked the forwarding (re-use) of port 5060 as they intend on the built in ATA to use that. This is not unique to Orange, or a device such as the Livebox, but to honest, I haven't actually heard of someone making this work (using a similar device), with 3CX. Maybe you''l be the first, but I don't give it much hope as the provider has probably "engineered" it to not work the way you want it to. They may even have a way of detecting that you have made modifications, trying to "get around" the ATA, which may be in violation of the terms of service.

In most cases, where a provider supplies a box, with FXS ports (dialtone), users will attach a gateway, then register that gateway as a trunk on the PBX. This is in the same manner, as would be done, with any PSTN line.

Thanks for your answer, in relation to what you say to set it as PSTN, my PC is a little "old man" and still incorporates a FXO entry, I connected a RJ11 hence cable to the FXS router, but not how to make 3cx detect it. Haber if someone throws me a cable. Thank you!

 

[leejor]

You would have to purchase a gateway, something like the (now discontinued) Linksys SPA-3102. there are other single line gateways around, do some searches.

 

[Sopock]

http://i65.tinypic.com/6f1uf6.png

Situation looks much more favorable than here where no softphones are allowed! It should be possible to convince device that 3CX is just generic softphone client.
I would compare INVITE from successful call in X-Lite(*106) with failed <sip:[email protected]:5060>

Are you trying to bypass the built in ATA and register directly with the provider?

intermediate step, it would be good to first add another account on ATA and register it to 3CX for internal calls :idea:

In most cases, where a provider supplies a box, with FXS ports (dialtone), users will attach a gateway, then register that gateway as a trunk on the PBX. This is in the same manner, as would be done, with any PSTN line.

other rare cases would be when users are not rich enough to buy outdated equipment and when they want to experience HD voice if available? For instance, eventually I will be able to send occasional fax through this provider without machine...

They may even have a way of detecting that you have made modifications, trying to "get around" the ATA, which may be in violation of the terms of service.

sometimes users may become pretty frustrated when they are forced to only use analog phones or to pay license fee for SD terrestrial.

 

[leejor]

If you can get other SIP devices to register with the box, there shouldn't really be a (technical) reason why 3CX can't. But, there may be other factors involved, that I'm not aware of. You may be able to find more information on a forum dedicated to the Livebox, and in particular, registering one type or another of PBX with it. Think of another VoIP PBX that is popular with hobbiests, search for that and Livebox.

It may very well be a simple setting that requires a change, or not, I can't say not having used the Livebox.

 

[erdia]

If you can get other SIP devices to register with the box, there shouldn't really be a (technical) reason why 3CX can't. But, there may be other factors involved, that I'm not aware of. You may be able to find more information on a forum dedicated to the Livebox, and in particular, registering one type or another of PBX with it. Think of another VoIP PBX that is popular with hobbiests, search for that and Livebox.

It may very well be a simple setting that requires a change, or not, I can't say not having used the Livebox.

I don't know.I test change VoIP Provider for SIP Trunk