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Tom Waterman

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Posts posted by Tom Waterman

  1. I have a snom M3 base and 3 handsets registered to the base. I have one extension that can not make or recieve any calls. If the Extension attempts to make a call I get an erroro message. Here are the 2 lines from the sip log on the base:

    0709140212 [N](09):Too many concurrent Calls by UA #2

    0709140212 [N](09):Call 415 rejected with cause 909 by peer (or system)

     

    There are NO calls on the base or in the pbx for that matter. I have the newest firmware of 2.02 on the phone. Please help.

     

    Tom

  2. It would be great if we can get a PCAP (Wireshark) trace of this when it happens. Maybe there is a problem with the jitter, or DTMF. We are shooting in the dark right now when it comes to this...

     

    Ok I have the recording and the PCAP trace. The PCAP is 4 megs zipped. What email address can I send it to?

     

    Tom

  3. No, you do not have to put the DID anywhere in the trunk settings. All you have to do is to put it under the extension's account number. Make sure that the carrier is sending the number what you have set on this filed (sometime 10 vs 11 digits can cause issues. So set the number that comes from the carrier).

     

    Yeah I did that. My carrier here at the office is sending 10 digits. I have put those 10 digits in my extension number but no dice. What do yo uneed from me to help troubleshoot this?

    Tom

  4. I had spoke with Kevin onsite about having a failover setup. One PBX would be the primary and somehow keep insync with the second. So if the first server goes down the secondary server would failover automatically. Can you tell me if this has been achieved and how?

     

    Tom

  5. Assigning DID is very simple. Edit each account and set the DID on the "Account Number(s)" field separated with a space. Ex: 4001 19787462777 (if you want to assign 19787462777 to extension 4001)

    Yeah I tried that and it does not work. Do I need to restart the service for this?

     

    Tom

  6. One thing is to make sure that the DID you assigned is the same as the one you receive from the carrier. A SIP log will tell you the details (whether you are receiving 7,10 or 11 digits on the incoming call)

     

    Is there a document that says how to set this up? I have about 25 users I need to do this for. Each user has their own direct dial DID and I need to make sure they go to the 4 digit extension that they already have assigned. I have also tried the hunt group and addidng the DID to the extension number like so 817 802748XXXX and it fails to redirect me to my extension when I call the number. I did look at the syslog and I am receiving 10 digits from the telco.

     

    Thank you in advance for the help.

  7. Is they're a limit to the length of cable in the signal quality?

    grade 5 cabeling with Cisco POE switch.

     

    If yes, what should be done if cable are 500 feets longs?

     

    In order to do a 500ft run you would need to put a switch/hub/repeater in the middle. Anything that will regenerate and retime the signal.

  8. Is there anyway the PBX would prevent a call from coming in? Here is my setupo. I have an older T1 that comes into an Audiocodes Mediant 1K. I can make 2 calls at the same time with no problem. On the 3rd call the caller receives a fast busy. It does not make any difference if it is 3 calls coming into the system or one in and 2 out or 3 out. It is the addition of the 3 call that causes this. Here is what I see in the Gateway.

     

    SIP/2.0 500 Line Unavailable<013><010>Via: SIP/2.0/UDP 10.130.1.236;branch=z9hG4bKac388051577<013><010>From:

     

    And then the reorder tone is played.

    Any help would be greatly appreciated.

     

    I have received a call back from audiocodes and they told me that it is th PBX stating the line is no available and not the gateway.

    Tom

  9. Is there anyway the PBX would prevent a call from coming in? Here is my setupo. I have an older T1 that comes into an AUdiocodes Mediant 1K. I can make 2 calls at the smae time with no problem. On the 3rd call the caller receives a fast busy. It does not make any difference if it is 3 calls coming intot he system or one in and 2 out or 3 out. It is the addition of the 3 call that causes this. Here is what I see in the Gateway.

     

    SIP/2.0 500 Line Unavailable<013><010>Via: SIP/2.0/UDP 10.130.1.236;branch=z9hG4bKac388051577<013><010>From:

     

    And then the reorder tone is played.

    Any help would be greatly appreciated.

     

    Tom

    I just got a response back from the gateway support folks and this disconnect is coming from the PBX side.

  10. Is there anyway the PBX would prevent a call from coming in? Here is my setupo. I have an older T1 that comes into an AUdiocodes Mediant 1K. I can make 2 calls at the smae time with no problem. On the 3rd call the caller receives a fast busy. It does not make any difference if it is 3 calls coming intot he system or one in and 2 out or 3 out. It is the addition of the 3 call that causes this. Here is what I see in the Gateway.

     

    SIP/2.0 500 Line Unavailable<013><010>Via: SIP/2.0/UDP 10.130.1.236;branch=z9hG4bKac388051577<013><010>From:

     

    And then the reorder tone is played.

    Any help would be greatly appreciated.

     

    Tom

  11. Go to the domain settings, at the bottom you can select the dial plan that should be provisioned (if available for the phone type).

     

    So I was able to get this working but let me tell you what happened so others may know the issue. When we originally installed the system we could not get PnP to work properly. However, with the lat couple of upgrades pnp is now functional, although I did not know this. So the customer had to reboot a couple of the phones and it pulled down the PnP dial plan which was set to 4 Digit North America. Because this client still has 7 digit dialing they would try to dial 358 and on the 4th digit it would dial. This would happen on any of thier local echanges. Ugg. So you would think the if you change the dial plan to "User Must Press Enter" and reboot the phone that this would fix the issue however it does not because onthis setting a dial plan xml file is not generated. Sooo, on the snom 370 I saved the settings and edited the current xml and reloaded it. This fixed the issue! Thanks to, Kevin, Kevin, and Pradeep for the help.

     

    Tom

  12. Did you previously provision a dialplan through PnP? If you turn that off later, then the phone will "stick" to the old plan. The only way to get it out of the phone is to either provision another plan or to factory-reset the phone.

     

    How do I provision another plan?

  13. I have a customer who has the snom 370 phones and they have a local exchange of 358. Until last week they could dial this local exchange no problem. I have made no changes to the PBX in a month at least and the snoms are running the newest firmware 7.3.14. Apparently last week some of the phones rebooted and auto provisioning actually kicked in. So now these phones when you dial 358 on the fourth digit it will auto dial. I can see this in the PBX wireshark which I have attached. This is a major issue as it is a PD and need immediate help.

  14. I ran the manual upgrade procedure following notes http://wiki.pbxnsip.com/index.php/Installi...#Manual_Upgrade but we were unable to dial out and received an error stating we had no credit left. I had to switch back to the old version till I could figure out what is wrong. Does anyone have any ideas?

     

    Thanks

    I have that problem with local calls. The customer can call a long distance number no problem but a local call it dials the first 4 digits and of course errors out.

  15. Hi everyone. I have acustomer that reports they can only make 2 outbound calls at a time and the 3rd call fails. It looks as though the call clears the PBX and hits the gateway but I am not sure. Anyone have any ideas?

     

    Also the same custome has a couple of snom phones 370 with the latest firmware that when you attempt to make a local call it treats it like it is a 4 digit extendion and it starts to dial.I have number guessing and auto dial turned off.

     

    Please help.

     

    Tom

  16. Maybe give version 2.02 a shot. Should be available now; apart from that maybe the distance or receiption is a problem. Although it is digital, maybe the device wants to reduce potential noice and lowers the volume. Just a wild guess :lol: ...

     

    I have updated to 2.02 I'll let you know!

     

    Tom

  17. Hello again, I have a user who has a snom m3 with firmware 1.22 and while on speaker phone the volume will fluctuate up and down. Has anyone seen this or heard of this before. I wish I would have known how crappy the speaker phone was before we ordered 50 of these things.

     

    Tom

     

    This sound quality issue also seems to happen when the user is not on speaker phone. If they are talking on the handset normally. All of them use the M3. Anyone else ever seen this? I have it on about 6 phones now.

    Please help.

  18. Hello again, I have a user who has a snom m3 with firmware 1.22 and while on speaker phone the volume will fluctuate up and down. Has anyone seen this or heard of this before. I wish I would have known how crappy the speaker phone was before we ordered 50 of these things.

     

    Tom

  19. Maybe just turn TLS off. In the email settings there is a flag called "Enable TLS", try turning it off.

    I tried that before posting. I get this error instead complaining about the authentication type. Also thier domain name is only localhost and thge IP address of the server the FQDN is not in there.

     

    2] 2009/04/20 15:13:32: Last message repeated 13 times

    [6] 2009/04/20 15:13:32: Sending CDR email to <twaterman@md-911.com>

    [8] 2009/04/20 15:13:33: SMTP: Connect to 10.130.1.27:25

    [8] 2009/04/20 15:13:33: SMTP: Received 220 mailserver@domain.org Microsoft ESMTP MAIL Service ready at Mon, 20 Apr 2009 15:13:33 -0500

     

    [8] 2009/04/20 15:13:33: SMTP: Send EHLO localhost

     

    [8] 2009/04/20 15:13:33: SMTP: Received 250-jascomx.jasco.org Hello [10.130.1.17]

    250-SIZE

    250-PIPELINING

    250-DSN

    250-ENHANCEDSTATUSCODES

    250-STARTTLS

    250-X-ANONYMOUSTLS

    250-AUTH NTLM

    250-X-EXPS GSSAPI NTLM

    250-8BITMIME

    250-BINARYMIME

    250-CHUNKING

    250-XEXCH50

    250 XRDST

     

    [8] 2009/04/20 15:13:33: SMTP: Send AUTH LOGIN

     

    [8] 2009/04/20 15:13:38: SMTP: Received 504 5.7.4 Unrecognized authentication type

     

    [5] 2009/04/20 15:13:38: SMTP Server returned 504

  20. Ok I have a very simple customer who just needs to get basic voicemail to email set up. They have exchange 2007 installed however the tech staff onsite are not familiar with it. I am trying to send a basic crd log to my email address and I get the following log entries. (domain name change to protect customer)'

     

    6] 2009/04/20 14:15:58: Sending CDR email to <twaterman@md-911.com>

    [8] 2009/04/20 14:15:58: SMTP: Connect to 10.130.1.27:25

    [8] 2009/04/20 14:15:58: SMTP: Received 220 mailserver@domain.org Microsoft ESMTP MAIL Service ready at Mon, 20 Apr 2009 14:15:57 -0500

     

    [8] 2009/04/20 14:15:58: SMTP: Send EHLO localhost

     

    [8] 2009/04/20 14:15:58: SMTP: Received 250-jascomx.jasco.org Hello [10.130.1.17]

    250-SIZE

    250-PIPELINING

    250-DSN

    250-ENHANCEDSTATUSCODES

    250-STARTTLS

    250-X-ANONYMOUSTLS

    250-AUTH NTLM

    250-X-EXPS GSSAPI NTLM

    250-8BITMIME

    250-BINARYMIME

    250-CHUNKING

    250-XEXCH50

    250 XRDST

     

    [8] 2009/04/20 14:15:58: SMTP: Send STARTTLS

     

    [8] 2009/04/20 14:15:58: SMTP: Received 220 2.0.0 SMTP server ready

     

    [8] 2009/04/20 14:15:58: SMTP: Send EHLO localhost

     

    [5] 2009/04/20 14:15:58: SMTP: Connection refused on 10.130.1.27:25

     

     

    Any help would be great!

     

    Tom

  21. ...I do think pbxnsip product would be more professional if this situation would be avoided...

     

    (no vm created for the 0 press)

    Although its not a huge thing.

     

    matt

    Yeah I would really like to see some sort of fix. I can even leave a message and hit 0 and I get the sound at the end of the message as well.

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