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asterisk_nicht_mehr

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Posts posted by asterisk_nicht_mehr

  1. Just as I suspected. this has absolutely no effect on the problem. I sent the call to a second extention which does not have a hunt group. The same thing happens. This does not have to do with the carrier. I called in using my mobile and the system still listed the area code stored in pbxnsip before the number of my handy.in this case:

     

    2011/01/19 22:12 021611716454xxx Fourty Two_SKpvt (42)(42) 00:06 PSTN Gateway 18

     

    The 02161 is coming from one of the fields in pbxnsip. Using my mobile I should have gotten either 01716454xxx or 1716454xxx. So where is the number coming from? Is is coming from the domain, or the trunk?

     

     

  2. Excuse me, but we need to think what could deliver the info to the report.

     

    2011/01/17 14:05 0216121116454715 72(40) PSTN Gateway 18 02161 is my city code then comes 211 the number from the incoming call from Düsseldorf

    2011/01/17 14:05 0216121116454715 72(40) 10:22 PSTN Gateway 18

     

    So the problem is still there and reproduceable. The problem must be in the way the system is set up. It only happens on incoming calls. Can you make any wild guesses? In this case it is coming on the hunt group.

     

    2011/01/17 13:32 021611737887422 Fourty Two_SKpvt (42)(42) 00:02 PSTN Gateway 18 02161 is my city code but 173 is the code for a mobile carrier. Ths number is coming through directly to the extention.

     

    What is constant is that my city code shows up consistantly on incoming calls received within Germany. The second point is that the number from the incoming caller ID is striping off the 0 before the 173 or 211. It must be a combination of info in a couple of fields.

     

    Think about it some more...perhaps it is a bug. The release level is 4.2.0.3958 (Win32).

     

    Give it some thought. Do I need to do a wireshark capture?

     

    Thanks.

     

     

     

     

    BTW, we double checked the double entry issue in the lab in Germany and we did not see the issue. We saw only one record.

  3. Excuse me, but we need to think what could deliver the info to the report.

     

    2011/01/17 14:05 0216121116454715 72(40) PSTN Gateway 18 02161 is my city code then comes 211 the number from the incoming call from Düsseldorf

    2011/01/17 14:05 0216121116454715 72(40) 10:22 PSTN Gateway 18

     

    So the problem is still there and reproduceable. The problem must be in the way the system is set up. It only happens on incoming calls. Can you make any wild guesses? In this case it is coming on the hunt group.

     

    2011/01/17 13:32 021611737887422 Fourty Two_SKpvt (42)(42) 00:02 PSTN Gateway 18 02161 is my city code but 173 is the code for a mobile carrier. Ths number is coming through directly to the extention.

     

    What is constant is that my city code shows up consistantly on incoming calls received within Germany. The second point is that the number from the incoming caller ID is striping off the 0 before the 173 or 211. It must be a combination of info in a couple of fields.

     

    Think about it some more...perhaps it is a bug. The release level is 4.2.0.3958 (Win32).

     

    Give it some thought. Do I need to do a wireshark capture?

     

    Thanks.

     

     

     

     

    BTW, we double checked the double entry issue in the lab in Germany and we did not see the issue. We saw only one record.

  4. Regarding the first one: the problem is the citycode coming in from the caller id being added to the area code in the General Setup. The first entry is from the Incoming Caller ID and the 2161 is from the area code field.

    Thanks but I used x to blockout the telephone numbers of people who called in.

     

    Thanks again

     

     

     

     

     

     

    The first one, I am not sure it was a copy & paste error on your part. I see that both entries have 0 in front. Apart from that it should show the area code / city code twice. We will double check this (in German lab :-))

     

    On the second one, do you know if the number comes to PBX in the INVITE message? Also, please see if hunt group setting "From-Header:" makes any difference.

  5. How should the From-Header be set? It presently has Calling-Party entered.

     

     

     

     

     

     

    The first one, I am not sure it was a copy & paste error on your part. I see that both entries have 0 in front. Apart from that it should show the area code / city code twice. We will double check this (in German lab :-))

     

    On the second one, do you know if the number comes to PBX in the INVITE message? Also, please see if hunt group setting "From-Header:" makes any difference.

    [/quote

  6. Hello everyone,

    Can you tell me why the call history would double entry the city code with one entry containing 0 and then without? I have 2161 in the area code under domain settings as the area code. I am in Europe,specifically Germany.

    2011/01/14 12:05 021612161xxxxxxx Fourty Two (42)(42) 00:05 PSTN Gateway 18

    2011/01/14 12:17 021612161xxxxxxx Fourty Two (42)(42) PSTN Gateway 18

     

    Secondly, because I had to set up my hunt group 72 with the incoming target number xxxxxx. I am getting an incoming call history that looks like this rather than incoming call details:

    2011/01/15 13:46 anonymous 72(40) 00:08 PSTN Gateway 18

    2011/01/15 13:46 anonymous 72(40) 00:08 PSTN Gateway 18

     

    This does not allow me to track who called inward because the system is not logging the incoming telephone number. Is there another way to set this up so I get the incoming number without loosing my hunt group?

     

    Thanks in advance.

  7. Here are the answers to your questions:

    4.2.0.3958 (Win32) Pbxnsip version

     

    * Check if that snom 370 is registered to the extension that you are expecting to ring.

    The Snom is set up under extentions as nr. 41 and the system states that it is registered

    * See if that extension is setup on the Hunt group's stage 1(if you want that phone to ring from the beginning).

    Both that extention and a second are in all three levels.

    * Verify if there are any night service flag is active on the hunt group.

    The night service was not set. I changed it.

     

    I found the solution. The solution is to put the common number on the hunt group rather than on a specific extention. When the call comes in it will automatically be assigned to both lines and roll off to the answering service in the end stage. That was a bit more elegant a solution than I had hoped for.

  8. Here are the answers to your questions:

    4.2.0.3958 (Win32) Pbxnsip version

     

    * Check if that snom 370 is registered to the extension that you are expecting to ring.

    The Snom is set up under extentions as nr. 41 and the system states that it is registered

    * See if that extension is setup on the Hunt group's stage 1(if you want that phone to ring from the beginning).

    Both that extention and a second are in all three levels.

    * Verify if there are any night service flag is active on the hunt group.

    The night service was not set. I changed it.

     

    The only way I can get a phone to ring is to enter it in under Extension 40 828752. The problem with that is, that alias only applies to one of my two extentions. Listing both with the same ANI does not work either.

  9. I am having problems re setting the Hunt Group after an update to a Patton 4652. I am getting the inbound calls to hit the mark but the Hunt Group is not ringing a Snom that is installed. It rings on a Siemens that is in the network (pbxnsip) but not on the Snom 370. What would someone need to do some diagnosis?

  10. 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.

  11. 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.

  12. 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?

  13. 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!

  14. 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?

  15. 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?

  16. 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?

  17. 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.

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