Jump to content

Help HELP NO SERVICE


asterisk_nicht_mehr
 Share

Recommended Posts

Hello,

 

This should be easy and was working before but has for some reason stopped. Country: Germany. Trunk will be ISDN to Deutsche Telecom. A typical 1 line BRI with 2 channels. Telefon numbers are preconfig as per the Telekom. All phones are registered. Outgoing telephone line is live. Gateway is also working so no problem there.

 

The Modem asigns DNS and has DHCP active.

 

I am having problems now with both outgoing and incoming. Pbxnsip ver. 3.0.1.3023 (Win32) Vigor2700 series Modem- 10 days ago there was a problem and the modem had to be restarted and when it did it reset the ip addresses. I relocated everything and thought I would be in the clear. I opened 5060-61 both TCP/UDP on the modem to pass it to the PBX. No other ports have been opened to the PBX on the Draytek. The PBX does not work either in the DNZ or behind the firewall. OK.

 

Of three phones I had one that worked: a Snom 370. I tried to copy the settings for it to the other but messed up a setting somehow now I have 3 lines that are not working. My girlfriend is having a party today and no one can call in. She must love me or maybe she is waiting until afterwards to kill me with the cake knife. If you deal with technology you can appreciate the seriousness of my need for help.

 

 

We are getting no dial tone. When people try to call us there is no ringing tone for them either.

On the trunk my settings look like this.

Sipgateway In and out bound

Domain and outbound proxy match

strict RTP is no

accept Redirect is no

Interpret SIP as telephone nr is on

prefix is empty

global is set to yes

trunk ani has the number of the ISDN number I assigned one phone (so we can speak on both channels at the same time)

Remote party Id is set to Remote party ID

No fallover

not secure

ICID empty

send to extention 40

assume call is from emply

ringback is Message 180

 

HELP HELP HELP she will be wakeing up soon.

Link to comment
Share on other sites

My girlfriend is having a party today and no one can call in. She must love me or maybe she is waiting until afterwards to kill me with the cake knife.

 

That's pretty serious!

 

If internal calls work fine, then we must have a problem with the DMZ and/or DNS. This sounds like the original setup had the DMZ set up in a slightly different way. Or maybe the new firewall has a different firmware version, and the forwarding works in a different way. Maybe it also has a logging feature that you can use to find out where the problem is.

 

Next time, also make a backup of the firewall. Every component can fail.

Link to comment
Share on other sites

That's pretty serious!

 

If internal calls work fine, then we must have a problem with the DMZ and/or DNS. This sounds like the original setup had the DMZ set up in a slightly different way. Or maybe the new firewall has a different firmware version, and the forwarding works in a different way. Maybe it also has a logging feature that you can use to find out where the problem is.

 

Next time, also make a backup of the firewall. Every component can fail.

 

What should I do now in what order?

Link to comment
Share on other sites

Check if the router sends SIP traffic to the PBX upon an incoming call. If that is the case, check the log of the PBX why it would reject that call. If there is no SIP traffic going to the PBX, check the router setup (again).

 

Again when I dial in using my mobile I get no tone. On the pbx there is no record of a call.

 

How should the router be set up differently?

Link to comment
Share on other sites

Again when I dial in using my mobile I get no tone. On the pbx there is no record of a call.

 

How should the router be set up differently?

 

From what I read in this topic, you had to replace your old router with a new one and since then íncoming calls don't work and more. That tells me that something in the setup of the router must have changed. Maybe it is something simple like the PBX has a new IP address and the DMZ settings must be adjusted accordingly. But it might also be a problem like the new router is suddenly SIP-aware and creating a compatibility nightmare. It is hard to say from here what the problem is. If I had access to the router, I would check the DMZ, the provided IP address (DHCP) and if the firmware is the same, and if there is something else suspicious. I would also check the log file of the router for messages.

Link to comment
Share on other sites

From what I read in this topic, you had to replace your old router with a new one and since then íncoming calls don't work and more. That tells me that something in the setup of the router must have changed. Maybe it is something simple like the PBX has a new IP address and the DMZ settings must be adjusted accordingly. But it might also be a problem like the new router is suddenly SIP-aware and creating a compatibility nightmare. It is hard to say from here what the problem is. If I had access to the router, I would check the DMZ, the provided IP address (DHCP) and if the firmware is the same, and if there is something else suspicious. I would also check the log file of the router for messages.

 

NO same Router. Did you look at the info I sent regarding the trunk? It could have something to do with the way the trunk is set up. Perhaps the way the code is being presented. Or is it stopping at the router?

Link to comment
Share on other sites

NO same Router. Did you look at the info I sent regarding the trunk? It could have something to do with the way the trunk is set up. Perhaps the way the code is being presented. Or is it stopping at the router?

 

Do you have a chance to see where the router sends packets that come to port 5060? Maybe there is another device in the network that also uses port 5060 and opens a connection to the Internet - and then that port 5060 is taken already by the other device, not the PBX. Many routers have a way of seeing how the ports are allocated.

 

If you are registering a trunk to a service provider, you will also see the real IP address and port in the Via header of the response.

 

[Did I mention I cannot wait for IPv6? No more of these NAT problems.]

Link to comment
Share on other sites

Do you have a chance to see where the router sends packets that come to port 5060? Maybe there is another device in the network that also uses port 5060 and opens a connection to the Internet - and then that port 5060 is taken already by the other device, not the PBX. Many routers have a way of seeing how the ports are allocated.

 

If you are registering a trunk to a service provider, you will also see the real IP address and port in the Via header of the response.

 

[Did I mention I cannot wait for IPv6? No more of these NAT problems.]

 

Those ports are directed to the PBX by the modem. They are not directed elsewhere. I have no problemwith the SIPgate connection. It is registered. My problem is with the ISDN lines not anything else. I do not know if the city code must be included as well. I still think it is a problem with the Trunk. If I do not make progress I can reinstall the Modem from scratch. Ugh!

Link to comment
Share on other sites

Those ports are directed to the PBX by the modem. They are not directed elsewhere. I have no problemwith the SIPgate connection. It is registered. My problem is with the ISDN lines not anything else. I do not know if the city code must be included as well. I still think it is a problem with the Trunk. If I do not make progress I can reinstall the Modem from scratch. Ugh!

 

Did you set the country code for the domain? If that is the case, the PBX tries to be smart about the numbers. If you put a "49" there and use the area code (e.g. "40"), then a number like 00494012345 will be interpreted as "12345". Try to use the DID numbers in the way you would dial them from the phone. Maybe you have a problem with the matching of the numbers. In this case, you should see the INVITE packet coming to the PBX with the number (do you? what do you see?).

Link to comment
Share on other sites

Did you set the country code for the domain? If that is the case, the PBX tries to be smart about the numbers. If you put a "49" there and use the area code (e.g. "40"), then a number like 00494012345 will be interpreted as "12345". Try to use the DID numbers in the way you would dial them from the phone. Maybe you have a problem with the matching of the numbers. In this case, you should see the INVITE packet coming to the PBX with the number (do you? what do you see?).

 

I will have time on Friday to sort the system out if you are around. If I may first describe some things.

 

When I try to dial in I get no tones at all. The caller calling in hears silence. An announcement from Telekom says that the party you are calling is not available.

 

I have the system set up with Ringback Message 180. Is there another setting that can cause that?

Secondly, I can make all calls to and from extentions internally. That means the PBX sees the phones and is directing them to the proper places. So internally there is not a problem with the DHCP or with DNS. Do you agree?

 

I have openned the 5060 for TCP/UDP and have directed them to the PBX. As a result, the Sipgate account I have for an incomng DID registers.

 

I do not want to direct calls outbound over that DID. I want to send them all over the 1 ISDN BRI line we have. When one dials outbound we get a dial tone. We get no ring tone but we hear after a few seconds a fast beep tone. At times we have had a tone that is a series of 3 rising tones.

 

I imagine it is the pbx that is generating the dial tone but if not, then it is getting through the NAT. If it is internally generated then Nat may still be the issue. What do you think?

Link to comment
Share on other sites

I will have time on Friday to sort the system out if you are around. If I may first describe some things.

 

When I try to dial in I get no tones at all. The caller calling in hears silence. An announcement from Telekom says that the party you are calling is not available.

 

I have the system set up with Ringback Message 180. Is there another setting that can cause that?

Secondly, I can make all calls to and from extentions internally. That means the PBX sees the phones and is directing them to the proper places. So internally there is not a problem with the DHCP or with DNS. Do you agree?

 

I have openned the 5060 for TCP/UDP and have directed them to the PBX. As a result, the Sipgate account I have for an incomng DID registers.

 

I do not want to direct calls outbound over that DID. I want to send them all over the 1 ISDN BRI line we have. When one dials outbound we get a dial tone. We get no ring tone but we hear after a few seconds a fast beep tone. At times we have had a tone that is a series of 3 rising tones.

 

I imagine it is the pbx that is generating the dial tone but if not, then it is getting through the NAT. If it is internally generated then Nat may still be the issue. What do you think?

 

I must admit that the routing to the ISDN Trunks is not clear to me from the online documentation. I understand to set up a tel:for each account. I understand that as the DID you mentioned. There is a provision in the system for establishing an ANI at the trunk and extention level. I am unclear if the information should be doubled or not.

 

In Trunk set up I am also unclear if the User should be set to blank, the outbound proxy, a name or something else. Thanks for you patience with me.

Link to comment
Share on other sites

I must admit that the routing to the ISDN Trunks is not clear to me from the online documentation. I understand to set up a tel:for each account. I understand that as the DID you mentioned. There is a provision in the system for establishing an ANI at the trunk and extention level. I am unclear if the information should be doubled or not.

 

In Trunk set up I am also unclear if the User should be set to blank, the outbound proxy, a name or something else. Thanks for you patience with me.

 

Hello again. I am home earlier than expected due to illness, so I am available to work with you earlier than the 16 Uhr 4oclock CET. I can now be available between 1340 till 16:30. 1:40 to 4:30 pm

 

I look forward to having your help. 10 days without the system is a long time. Call me on +49 171 645-4955 so that can get ready by the system.

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...