Jump to content

pbxuser911

Members
 View Profile  See their activity

  • Posts

    498

  • Joined

  • Last visited

 Content Type 

Posts posted by pbxuser911

  1. Enable 911

    in ITSP's

    Posted

    this is was i was able to grabw hen they tryed calling us back log level is at 5

    they asked me to add a prefix of number 1, since thats another option on how they know which domain had called to them.

     

     

     

    [5] 2009/01/08 14:14:10: SIP Rx udp:208.71.179.144:5060:

    INVITE sip:12486@xx.xx.xx.xxSIP/2.0

    Via: SIP/2.0/UDP 208.71.179.144:5060;branch=z9hG4bK14ad99f6;rport

    From: "5147452143" <sip:5147452143@208.71.179.144>;tag=as79f448b4

    To: <sip:12486@xx.xx.xx.xx>

    Contact: <sip:5147452143@208.71.179.144>

    Call-ID: 3af6eb4979dbf4a93096aa3f1ebf933d@208.71.179.144

    CSeq: 102 INVITE

    User-Agent: 911Enable SBC4

    Max-Forwards: 70

    Date: Thu, 08 Jan 2009 19:16:25 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Supported: replaces

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=root 4386 4386 IN IP4 208.71.179.144

    s=session

    c=IN IP4 208.71.179.144

    t=0 0

    m=audio 14860 RTP/AVP 0 8 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=silenceSupp:off - - - -

    a=ptime:20

    a=sendrecv

     

    [5] 2009/01/08 14:14:10: SIP Tx udp:208.71.179.144:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 208.71.179.144:5060;branch=z9hG4bK14ad99f6;rport=5060

    From: "5147452143" <sip:5147452143@208.71.179.144>;tag=as79f448b4

    To: <sip:12486@xx.xx.xx.xx>;tag=64acef8041

    Call-ID: 3af6eb4979dbf4a93096aa3f1ebf933d@208.71.179.144

    CSeq: 102 INVITE

    Content-Length: 0

     

     

    [5] 2009/01/08 14:14:10: SIP Tx udp:208.71.179.144:5060:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/UDP 208.71.179.144:5060;branch=z9hG4bK14ad99f6;rport=5060

    From: "5147452143" <sip:5147452143@208.71.179.144>;tag=as79f448b4

    To: <sip:12486@xx.xx.xx.xx>;tag=64acef8041

    Call-ID: 3af6eb4979dbf4a93096aa3f1ebf933d@208.71.179.144

    CSeq: 102 INVITE

    Contact: <sip:12486@xx.xx.xx.xx:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: PBX/3.1.1.3110

    Content-Length: 0

     

     

    [5] 2009/01/08 14:14:10: Last message repeated 2 times

    [5] 2009/01/08 14:14:10: SIP Rx udp:208.71.179.144:5060:

    ACK sip:12486@xx.xx.xx.xxSIP/2.0

    Via: SIP/2.0/UDP 208.71.179.144:5060;branch=z9hG4bK14ad99f6;rport

    From: "5147452143" <sip:5147452143@208.71.179.144>;tag=as79f448b4

    To: <sip:12486@xx.xx.xx.xx>;tag=64acef8041

    Contact: <sip:5147452143@208.71.179.144>

    Call-ID: 3af6eb4979dbf4a93096aa3f1ebf933d@208.71.179.144

    CSeq: 102 ACK

    User-Agent: 911Enable SBC4

    Max-Forwards: 70

    Content-Length: 0

  2. SPA 962 Dial Plan

    in Linksys Phones

    Posted

    I removed the * codes and it works fine, but as soon as i do restore the phone to the factory setting, it gets back its * code, so in the configuration file i set the following,

     

    <Call_Return_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Blind_Transfer_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Call_Back_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Call_Back_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Cfwd_All_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Cfwd_All_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Cfwd_Busy_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Cfwd_Busy_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Cfwd_No_Ans_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Cfwd_No_Ans_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <CW_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <CW_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <CW_Per_Call_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <CW_Per_Call_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Block_CID_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Block_CID_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Block_CID_Per_Call_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Block_CID_Per_Call_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Block_ANC_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Block_ANC_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <DND_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <DND_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Secure_All_Call_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Secure_No_Call_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Secure_One_Call_Act_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Secure_One_Call_Deact_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Paging_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Call_Park_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Call_Pickup_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Call_UnPark_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Group_Call_Pickup_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Media_Loopback_Code group="Regional/Vertical_Service_Activation_Codes" />

    <Referral_Services_Codes group="Regional/Vertical_Service_Activation_Codes" />

    <Feature_Dial_Services_Codes group="Regional/Vertical_Service_Activation_Codes" />

    <Service_Annc_Base_Number group="Regional/Vertical_Service_Announcement_Codes" />

    <Service_Annc_Extension_Codes group="Regional/Vertical_Service_Announcement_Codes" />

     

     

    but it still adds the codes back

  3. SPA 962 Dial Plan

    in Linksys Phones

    Posted

    Has anyone found a good dial plan to use with the SPA962 phone? right now when i dial *67 it doesnt block the caller ID, it just says enter number: and when i do so the caller ID is still not blocked

    but on a GrandStream phone wheni dial *67 it says : yoru caller id will be blocked on outgoing calls

     

    it seems that teh linksys phone has its own sets of codes that will cause it not to send the correct code to the PBXnSIP system?

  5. Enable 911

    in ITSP's

    Posted

    here is the log on the system whent hey try to call us:

     

    [4] 2009/01/06 13:35:43: SIP Rx udp:208.71.179.138:5060:

    INVITE sip:2485@xx.xx.xx.xxSIP/2.0

    Via: SIP/2.0/UDP 208.71.179.138:5060;branch=z9hG4bK37acdea7;rport

    From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as04e6bed8

    To: <sip:2485@xx.xx.xx.xx>

    Contact: <sip:5147452143@208.71.179.138>

    Call-ID: 35cf8b615939486c1f74b33709f0dc6b@208.71.179.138

    CSeq: 102 INVITE

    User-Agent: 911Enable SBC1

    Max-Forwards: 70

    Date: Tue, 06 Jan 2009 18:37:38 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Supported: replaces

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=root 4340 4340 IN IP4 208.71.179.138

    s=session

    c=IN IP4 208.71.179.138

    t=0 0

    m=audio 15842 RTP/AVP 0 8 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=silenceSupp:off - - - -

    a=ptime:20

    a=sendrecv

     

    [4] 2009/01/06 13:35:43: SIP Tx udp:208.71.179.138:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 208.71.179.138:5060;branch=z9hG4bK37acdea7;rport=5060

    From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as04e6bed8

    To: <sip:2485@xx.xx.xx.xx>;tag=f3aa4205c9

    Call-ID: 35cf8b615939486c1f74b33709f0dc6b@208.71.179.138

    CSeq: 102 INVITE

    Content-Length: 0

     

     

    [4] 2009/01/06 13:35:43: SIP Tx udp:208.71.179.138:5060:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/UDP 208.71.179.138:5060;branch=z9hG4bK37acdea7;rport=5060

    From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as04e6bed8

    To: <sip:2485@xx.xx.xx.xx>;tag=f3aa4205c9

    Call-ID: 35cf8b615939486c1f74b33709f0dc6b@208.71.179.138

    CSeq: 102 INVITE

    Contact: <sip:2485@xx.xx.xx.xx:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: PBX/3.1.1.3110

    Content-Length: 0

     

     

    [4] 2009/01/06 13:35:43: Last message repeated 2 times

    [4] 2009/01/06 13:35:43: SIP Rx udp:208.71.179.138:5060:

    ACK sip:2485@xx.xx.xx.xxSIP/2.0

    Via: SIP/2.0/UDP 208.71.179.138:5060;branch=z9hG4bK37acdea7;rport

    From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as04e6bed8

    To: <sip:2485@xx.xx.xx.xx>;tag=f3aa4205c9

    Contact: <sip:5147452143@208.71.179.138>

    Call-ID: 35cf8b615939486c1f74b33709f0dc6b@208.71.179.138

    CSeq: 102 ACK

    User-Agent: 911Enable SBC1

    Max-Forwards: 70

    Content-Length: 0

  6. When a call approaches the head of the queue

    in Agent Group Setup

    Posted

    When i set up an agent group

    i want the option that after 1 minute or 2 minutes or 45 seconds, the call should get transferred to another extension which is the mailbox, (would be nice to have a mail box for a agent group etc)

    where do i set that?

    if i set any of the: After hearing ringback for (s) ...

    ... include the following additional agents (e.g. "41 42 43"):

    After hearing ringback for (s) ...

    ... redirect the call to the destination (e.g. "73"):

    If the caller already waited longer than (s) ...

    ... redirect to the destination (e.g. "73"):

    the call doesnt get transferred while its in the que after setting it to 45 seconds

  7. Stress Testing

    in General Setup

    Posted

    thats how i load test.

     

    I should add that later on i get this message from pbxnsip:

     

    The call from sip:99@xxxxxxx.com:5060 to

    sip:sipp@xxxxxxx.com:5069 has been disconnected because of media timeout (120 seconds), 0/6000 packets have been received/sent

     

    i feel like im missing something in my scenario to disconneced the rtp stream correctly.

  8. Stress Testing

    in General Setup

    Posted

    thats how i load test.

     

    I was able to test using sipp without pcap_play fine but when i play an audio pcap I get messages like this from sipp:

     

    "Discarding message which can't be mapped to a known SIPp call"

     

    Any ideas?

  12. Overhead Paging

    in Linksys Phones

    Posted

    has anyone used a SPA 962 or SPA942 phone with the paging option?

    if a customer is on the phone with someone, a page comes in,the call gets placed on hold so the page comes in

     

    what should i look for to change so the call doesnt get placed on hold?

  18. this extension number does not exist

    in Auto Attendant Setup

    Posted

    happy new years to all

     

    ive ran into a problem that when in the auto attendant and you press a wrong extension then the pbx will say "this extension number does not exist" and then stays silent till the right extension is pressed, is there anyway to set it up that after announcing it, it should return to the last stage it was it, where in this case was the auto attendant?

    or maybe give us the option to tell the system what to do?

  19. Calls Declined

    in ITSP's

    Posted

    ill let TSG know about it

    as far as the sound file missing, i renamed the NOISE (comfort tone) sound so it doesnt play

    we find it not to work best in our set up

     

    thanks for the reply and bringing to my attention what you found besides the main issue i posted

  20. Calls Declined

    in ITSP's

    Posted

    Ok here is what I figured out the problem was, but need helping finding a way around it

     

    We have the name of the extensions W: Office 1

    the termination provider has declined it due to the :, once I remove : everything works good..

     

    Would anyone know how do i send the caller ID in quotations? supposedly that should work even there is a :

  21. Calls Declined

    in ITSP's

    Posted

    i try making a call using a certain trunk, the call wont connect and i get a declined error on the phone

    but when making the same call from a 2nd extension on the same domain and the same phone, it works

     

    ***Note all IP address, the proxy server Domain name, Phone numbers that i chose to keep private for the customer, has been replaced with, xx.xx.xxx.xxx companya.mydomain.com 9784256666***

    the phone i tryed calling was 19785551212

     

    3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

    INVITE sip:19785551212@companya.mydomain.comSIP/2.0

    Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-fe7f0737

    From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com>

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 101 INVITE

    Max-Forwards: 70

    Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

    Expires: 240

    User-Agent: Linksys/SPA962-6.1.3(a)

    Content-Length: 395

    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

    Supported: replaces

    Content-Type: application/sdp

     

    v=0

    o=- 30452 30452 IN IP4 192.168.1.106

    s=-

    c=IN IP4 xx.xx.xxx.xxx

    t=0 0

    m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:2 G726-32/8000

    a=rtpmap:4 G723/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:18 G729a/8000

    a=rtpmap:96 G726-40/8000

    a=rtpmap:97 G726-24/8000

    a=rtpmap:98 G726-16/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:30

    a=sendrecv

     

    [3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-fe7f0737

    From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 101 INVITE

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-fe7f0737

    From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 101 INVITE

    User-Agent: pbx/3.1.1.3110

    WWW-Authenticate: Digest realm="companya.mydomain.com",nonce="bc93d1c2f115fdfa416d8bef1fd53b2e",domain="sip:19785551212@companya.mydomain.com",algorithm=MD5

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

    ACK sip:19785551212@companya.mydomain.comSIP/2.0

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-fe7f0737

    From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 101 ACK

    Max-Forwards: 70

    Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

    User-Agent: Linksys/SPA962-6.1.3(a)

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

    INVITE sip:19785551212@companya.mydomain.com.com SIP/2.0

    Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-10e069bb

    From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com>

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 102 INVITE

    Max-Forwards: 70

    Authorization: Digest username="107",realm="companya.mydomain.com",nonce="bc93d1c2f115fdfa416d8bef1fd53b2e",uri="sip:19785551212@companya.mydomain.com",algorithm=MD5,response="5e3fe565d38eb9b0719748db7f5584a1"

    Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

    Expires: 240

    User-Agent: Linksys/SPA962-6.1.3(a)

    Content-Length: 395

    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER

    Supported: replaces

    Content-Type: application/sdp

     

    v=0

    o=- 30452 30452 IN IP4 192.168.1.106

    s=-

    c=IN IP4 xx.xx.xxx.xxx

    t=0 0

    m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:2 G726-32/8000

    a=rtpmap:4 G723/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:18 G729a/8000

    a=rtpmap:96 G726-40/8000

    a=rtpmap:97 G726-24/8000

    a=rtpmap:98 G726-16/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:30

    a=sendrecv

     

    [3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-10e069bb

    From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 102 INVITE

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:5060:

    INVITE sip:19785551212@xx.xx.xxx.xxx;user=phone SIP/2.0

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:5060;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport

    From: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>;tag=648644319

    To: <sip:19785551212@xx.xx.xxx.xxx;user=phone>

    Call-ID: 7b24ce66@pbx

    CSeq: 6743 INVITE

    Max-Forwards: 70

    Contact: <sip:9784256666@xx.xx.xxx.xxx:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbx/3.1.1.3110

    P-Asserted-Identity: "W: Office 1" <sip:9784256666@companya.mydomain.com ;user=phone>

    Content-Type: application/sdp

    Content-Length: 300

     

    v=0

    o=- 1344194633 1344194633 IN IP4 xx.xx.xxx.xxx

    s=-

    c=IN IP4 xx.xx.xxx.xxx

    t=0 0

    m=audio 41368 RTP/AVP 0 8 3 18 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [3] 2008/12/31 22:35:09: Could not open WAV file audio_moh/noise.wav

    [3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

    SIP/2.0 183 Ringing

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:1025;branch=z9hG4bK-10e069bb

    From: "107" <sip:107@companya.mydomain.com>;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com>;tag=edf6f65045

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 102 INVITE

    Contact: <sip:107@74.206.239.196:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbx/3.1.1.3110

    Content-Type: application/sdp

    Content-Length: 289

     

    v=0

    o=- 1165514095 1165514095 IN IP4 xx.xx.xxx.xxx

    s=-

    c=IN IP4 xx.xx.xxx.xxx

    t=0 0

    m=audio 19926 RTP/AVP 0 8 18 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=sendrecv

     

    [3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:5060:

    SIP/2.0 100 Giving a try

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:5060;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport=5060

    From: "W: Office 1" <sip:9784256666@companya.mydomain.com ;user=phone>;tag=648644319

    To: <sip:19785551212@xx.xx.xxx.xxx;user=phone>

    Call-ID: 7b24ce66@pbx

    CSeq: 6743 INVITE

    Server: OpenSIPS (1.4.2-notls (x86_64/linux))

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: Could not open WAV file audio_moh/noise.wav

    [3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:5060:

    SIP/2.0 603 Declined

    Via: SIP/2.0/UDP 74.206.239.196:5060;received=74.206.239.196;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport=5060

    From: "W: Office 1" <sip:9784256666@companya.mydomain.com ;user=phone>;tag=648644319

    To: <sip:19785551212@xx.xx.xxx.xxx;user=phone>;tag=as6b435cf2

    Call-ID: 7b24ce66@pbx

    CSeq: 6743 INVITE

    User-Agent: TSG_Global_GW

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Supported: replaces

    Contact: <sip:19785551212@xx.xx.xxx.xxx>

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:5060:

    ACK sip:19785551212@xx.xx.xxx.xxx;user=phone SIP/2.0

    Via: SIP/2.0/UDP xx.xx.xxx.xxx:5060;branch=z9hG4bK-0d16d66d82498c26250e21c8181d4eb3;rport

    From: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>;tag=648644319

    To: <sip:19785551212@69.25.128.195;user=phone>;tag=as6b435cf2

    Call-ID: 7b24ce66@pbx

    CSeq: 6743 ACK

    Max-Forwards: 70

    Contact: <sip:7183840099@xx.xx.xxx.xxx:5060;transport=udp>

    P-Asserted-Identity: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: SIP Tx udp:xx.xx.xxx.xxx:1025:

    SIP/2.0 603 Declined

    Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-10e069bb

    From: "107" <sip:107@companya.mydomain.com >;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com >;tag=edf6f65045

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 102 INVITE

    Contact: <sip:107@xx.xx.xxx.xxx:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbx/3.1.1.3110

    Content-Length: 0

     

     

    [3] 2008/12/31 22:35:09: SIP Rx udp:xx.xx.xxx.xxx:1025:

    ACK sip:19785551212@companya.mydomain.com SIP/2.0

    Via: SIP/2.0/UDP 71.58.196.187:1025;branch=z9hG4bK-10e069bb

    From: "107" <sip:107@companya.mydomain.com >;tag=52f8d1546848e82fo2

    To: <sip:19785551212@companya.mydomain.com >;tag=edf6f65045

    Call-ID: 25c3fff8-6b34e9db@192.168.1.106

    CSeq: 102 ACK

    Max-Forwards: 70

    Authorization: Digest username="107",realm="companya.mydomain.com ",nonce="bc93d1c2f115fdfa416d8bef1fd53b2e",uri="sip:19785551212@companya.mydomain.com ",algorithm=MD5,response="5e3fe565d38eb9b0719748db7f5584a1"

    Contact: "107" <sip:107@xx.xx.xxx.xxx:1025>

    User-Agent: Linksys/SPA962-6.1.3(a)

    Content-Length: 0

  23. choppy audio in IVR

    in General Setup

    Posted

    I called into my system and the IVCR audio was very choppy,

    we restarted the server and now runs fine

    any clues what might of caused it?

    friday we were runnign SIPP (stress tester), could it be we left it runnign and never shut it down properly?

  24. Logging

    in General Setup

    Posted

    When setting up the log file name under the loggin settings

    if you use log$txt it will create the log file based on todays date, but i also read that it will delete any old files

    is there somethign else i can put in there that will create a new log file daily based on the date and it will also leave the old log file?

     

    thanks

×
×
  • Create New...