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reco

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Posts posted by reco

  5. troubleshooting dropped calls

    in General Setup

    Posted

    hi there,

     

    wants the best way to trouble shoot dropped calls?

     

    i have people complaining about dropped calls. i see the following in my logs

     

    [5] 20111102110243: BYE Response: Terminate 57013b3cc783-98y2n65yo8ry

     

    and this

     

     

    Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-de8a3ad89b13788879a54f5777f70b62;rport=5061
From: <sip:8004928468@domain.com;user=phone>;tag=0425310071
To: "Coleen Mac Queen" <sip:21@domain.com>;tag=pg3pm9np15
Call-ID: 57013b3cc783-98y2n65yo8ry
CSeq: 2314 BYE
Contact: <sip:21@10.0.24.111:4102;transport=tls;line=ljhsjt2p>;reg-id=1
User-Agent: snom870/8.4.32
RTP-RxStat: Total_Rx_Pkts=53264,Rx_Pkts=53264,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=53291,Tx_Pkts=53293,Remote_Tx_Pkts=53232
Content-Length: 0

     

     

    how can i denitrify where the issue is?

     

    thanx

  6. auto provision 2 sip accounts

    in Plug and Play

    Posted

    hi there,

     

    we had our pbxnsip on two different public ip addresses. some time one provider does some maintenance work and the line is down. is there a way to provision two sip profiles and use the 2nd as a backup in case the server of the first profile is not available?

     

    thanx

  9. modifying sip invige

    in General Setup

    Posted

    hi there,

     

    my trunk provider voxbeam is asking me to remove to 

    user=phone

    from the to line.

     

    change

     

    To: <sip:1212xxxxxxx@sbc.voxbeam.com;user=phone>;tag=VBSBC.1948.2045

     

    into

     

    To: <sip:1212xxxxxxx@sbc.voxbeam.com>;tag=VBSBC.1948.2045

     

    how can i do that?

    reco

  11. 404 not found on some calls after upgrade from version 3 to 4

    in General Setup

    Posted

    Does this user has any button configured on button 1? When you dial something, the phone sends the "seize" to the PBX if the button 1 configured to do co-line. This is an undesirable condition and can cause this issue.

     

    To verify this, you can modify the button configuration from "button 1" and reboot the phone (reboot is not really needed, but it will cleanup any old settings set on the buttons).

     

    this could be the issue. i deleted all the co lines from all trunks and domains. that solved the issue for now.

    will try to bring them back.

     

    if this is the case what should i do with button 1? just no config?

     

    x

  12. 404 not found on some calls after upgrade from version 3 to 4

    in General Setup

    Posted

    It could be that the user has seized a CO-line when placing the call (why is another question). Anyway, I remember there were some situations where the CO line could get stuck "on", but as far as I remember the situation goes away after the next reboot or after the seizure timeout (usually 60 seconds).

     

    i am actually tempted to track the concept of co-lines totally and use park orbits instead.

    they are trunk independent. what do you think?

     

    would be cool if there is a way to get rid of the feature codes in the dialed list and the announcement that a call was parked in the orbit #

     

    x

  16. 404 not found on some calls after upgrade from version 3 to 4

    in General Setup

    Posted

    i am monitoring my dial plans.

     

    seems rules which should match are skipped cause of co lines?

    can somebody explain me the reason for this?

    i would expect the pbx to send the call to trunk: voxbeam_js

     

     

    [8] 20111010171436: To is <sip:12129960700@johnsheeley.com;user=phone>, user 0, domain 5

    [8] 20111010171436: From user 20

    [8] 20111010171436: Call state for call object 455: idle

    [7] 20111010171436: set_codecs: for 70b7263c8244-u3zz195hutuu codecs "", codec_preference count 7

    [9] 20111010171436: Dialplan: Evaluating !^311!sip:12126399675@\r;user=phone!i against 2129960700@johnsheeley.com

    [9] 20111010171436: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 2129960700@johnsheeley.com

    [7] 20111010171436: Skipping pattern match because CO-line is not available for trunk voxbeam_js

     

     

    my dial plan:

    51;icall-domestic-sheeley;;^311;12126399675
101;voxbeam_js;;*;
400;AmericanVOIP domestic JS;;^([0-9]{10})@.*;"sip:1\1@\r;user=phone"
401;AmericanVOIP domestic JS;;^1([0-9]{10})@.*;"sip:1\1@\r;user=phone"
403;AmericanVOIP international JS;;^011([0-9]*)@.*;"sip:011\1@\r;user=phone"
500;icall-domestic-sheeley;;^([0-9]{10})@.*;"sip:1\1@\r;user=phone"
501;icall-domestic-sheeley;;^1([0-9]{10})@.*;"sip:1\1@\r;user=phone"
502;icall-international-sheeley;;^011([0-9]*)@.*;"sip:011\1@\r;user=phone"

     

    trunk `voxbeam_js` has no co lines

     

    any idea whats going on?

     

    thanx

  17. 404 not found on some calls after upgrade from version 3 to 4

    in General Setup

    Posted

    Hmm. It is not very clear. Do you have trunk logging turned on? If it would associate the call with a trunk, it would say so in the log. And if it would not associate the call with a trunk and the from-User is not known, then it would also say so. Otherwise, you would see that it tries to go through a dialplan, but again nothing in the log. For for now, I think the "bug" must be in the logging part...

     

    i think i found the issue.

     

    with version 3 i used to add a `Try Loopback` with pattern `*` Replacement `` (empty) in the beginning of a dial plan to enable inter domain calling followed by a trunk.

     

    dial plan csv:

     

    50;*;;*;
200;voxbeam_nex9;;*;

    with the loopback i have the issue. once i remove it seems to work fine:

     

    working dial plan:

     

    200;voxbeam_nex9;;*;

     

    reco

  19. 404 not found on some calls after upgrade from version 3 to 4

    in General Setup

    Posted

    hi there,

     

    Do you have a domain with the name "domain.com"? Configured on your PBX?

     

    nope i just replaced my domain with domain.com

     

    Did you set a country code for the domain? If you set it to "1", all numbers in the dialplan will be presented as 10-digit numbers (xxxxxxxxxx or 011yxz). Make sure that the dialplan does that.

     

    i have country code set to: 1

    area code to : 212

    phone number: 212 333 5555

     

    If you have only one domain running on your PBX, you can also add the name "localhost" to the list of domains, this acts like a wildcard

     

    i have multiple domains so not localhost configured

     

    Do you have a account with the name "12" on your system?

     

    yes absolutely. also i do have an extension 12 in other domains.

     

    Does this account have a dial plan assigned either on account level or domain level that allows dialling this number?

     

    account has a domain default dial plan

     

    I guess the call does never make it to the trunk level, looking at the log. Right?

     

    yes looks like.

     

     

    any suggestions?

  20. 404 not found on some calls after upgrade from version 3 to 4

    in General Setup

    Posted

    hi there,

     

    on some calls i am getting 404 not found and i cannot figure out why.

    any idea?

     

    thanx

     

     

     

    SIP/2.0 100 Trying

    Via: SIP/2.0/TLS 10.0.24.138:4993;branch=z9hG4bK-69lcg8nopbbd;rport=4993

    From: "First Last" <sip:12@domain.com>;tag=koerzr4mij

    To: <sip:2223335555@domain.com;user=phone>;tag=0a70de7b9f

    Call-ID: 1d74263c345d-dg32cs7hcuwn

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [8] 20111010131442: Incoming call: Request URI sip:2223335555@domain.com;user=phone, To is <sip:2223335555@domain.com;user=phone>

    [8] 20111010131442: Set the To domain based on From user 12@domain.com

    [9] 20111010131442: SIP Tx tls:10.0.24.138:4993:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/TLS 10.0.24.138:4993;branch=z9hG4bK-69lcg8nopbbd;rport=4993

    From: "First Last" <sip:12@domain.com>;tag=koerzr4mij

    To: <sip:2223335555@domain.com;user=phone>;tag=0a70de7b9f

    Call-ID: 1d74263c345d-dg32cs7hcuwn

    CSeq: 1 INVITE

    Contact: <sip:12@10.0.24.2:5061;transport=tls>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.2.1.4025

    Content-Length: 0

  24. rewrite from header on incoming call

    in Trunk Setup

    Posted

    is there a way to include some information which shows the user form which trunk the call came from?

     

    background info:

     

    my client has a private line which rings only one extension. he wants so see on the incoming call that it is that private line.

     

    i am using snom 870 and pbxnsip 3.4

     

    thanx

  25. transfer to busy extension

    in Call Treatment

    Posted

    i hear you,

     

    i did setup my virtual keys to monitor all extensions. when i have a call on hold its really hard to get to the virtual key menu.

    is there a better way to do it?

     

    my client has a receptionist which is transferring all the calls she/he would need to know if that person is on a call or available.

     

     

    reco

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