Jump to content

lirees

Members
 View Profile  See their activity

  • Posts

    34

  • Joined

  • Last visited

 Content Type 

Posts posted by lirees

  1. manage sf from cell phone

    in Using Cell Phones

    Posted

    very strange,

    i have inserted two cell phone number in two different extensions and in the log of pbx the numbers match at 100%.

    in fact in the "form" field of sip log when i call the pbx from a cell phone appears the name of the associated exstension, but the personal virtual assistan don't work

     

    INVITE sip:203@172.16.10.15:5060 SIP/2.0

    Via: SIP/2.0/UDP 172.16.10.3:5060;branch=z9hG4bK-99b7558304963810085ba0e25f4b561e;rport

    From: "Cordless" <sip:333xxxx305@172.16.10.3;user=phone>;tag=551319297

    To: <sip:03621xxx801@127.0.0.1;user=phone>

    Call-ID: 454a6904@pbx

    CSeq: 29660 INVITE

    Max-Forwards: 70

    Contact: <sip:203@172.16.10.3:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snomONE/4.5.0.1075 Delta Aurigids

    Content-Type: application/sdp

    Content-Length: 382

     

    v=0

    o=- 1760881727 1760881727 IN IP4 172.16.10.3

    s=-

    c=IN IP4 172.16.10.3

    t=0 0

    m=audio 60730 RTP/AVP 18 3 0 8 2 9 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:3 gsm/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

  2. manage sf from cell phone

    in Using Cell Phones

    Posted

    If the PBX has the extension associated with the cell phone, it should be actually quite simple. Call into the PBX (the private virtual assistant will offer you the special menu), then go to the auto attendant and then then dial the service flag number.

     

    If you need to dial a star code, you can call into your mailbox (PIN code), then in the main menu you can dial star codes. For example *80 can be used to set redirection for auto attendants, hunt groups and other account types (http://wiki.snomone.com/index.php?title=Call_Forwarding#Set_Night_Mode_for_Domain_Accounts_.28.2A80.29). This does not use service flags, but is also useful in cases you are out of the office, for example on snow days.

     

     

    it would be a perfect solution but I have any problem with the configuration of the pbx

     

    i have insert in the field "Cell phone number" of the tab "redirections" my cell number and changed "When this cell phone calls the PBX:" in "offer personal virtual assistant", but when i call the number of pbx from my cell phone i don't hear the special menu. So i have try to connect directly the line to the extension without passing to hunt group but nothing has changed.

     

    where i wrong ??

     

    thanks

  7. listen recorded message with dial code

    in Auto Attendant Setup

    Posted

    Is this the AA->IVR or IVR node account? If it is the IVR node, you can dial the node account number from any phone and it should play the the recorded message.

     

    this is the IVR node but if i call the node account i listen only the default message. I'll explain ....

    i record del message from the phone usising the dial code ( example *95530*1 or *98530*2 etc ) and i would like also hear these messages from the phone, there is a dial code to do this ???

  12. transfer call with siemens s865ip

    in VoIP Phones

    Posted

    i have a problem when i try to transfer a call from siemesn s865ip to another telephone

     

    this is the scenario :

    the gigaset answer the call

    options button

    select "external call"

    insert the number of extension ( snom320 )

    options button and i select "call transfert"

     

    the display of the gigaset show me "call transferred" but the call drop, if i try to transfer the call using the dial code *77ext the call is transferred without any problems

     

    thanks

  13. forwarding to Cell Phone on a different stages

    in Using Cell Phones

    Posted

    Why don't you use the "press 1 to accept the call" for the cell phones? in this case, don't put PSTN numbers into the hunt group, just the extensions. Then check that hunt groups may also call the extension's cell phone and tell the PBX to wait for the "1" confirmation tone from the cell phone. All set, even if the call goes to the cell phones mailbox it will still for connect such a call.

     

    great ... it worked !!!

     

    thanks

  15. connect remote office through vpn

    in General Setup

    Posted

    I do not know if it's correct but i solved by changing the configuration of the both trunk in this way :

    Accept Redirect: yes

    Assume that call comes from user: 203 for office1 and 303 for office2

     

    the extension 203 and 303 are a dummy user, in this way i can call form the office1 through the pstn and voip line of the office2 and viceversa

     

    now i should check with the blf of the snom320 in the office1 the status of the telephon in the office2, is it possible??? can i check also the sla ??

     

    thanks

  16. connect remote office through vpn

    in General Setup

    Posted

    is not a typo error, you're right, the ip of the office2 is 192.168.1.50 i have configure the trunk with the wrong ip .

    now i call the extension without problem but if i try to call a external numer form the exstension of office2 through the line of the office1 i give : 404 Not Found

     

    could be a problem of the dial plan ??

     

    DP office1

    pref 70

    trunk office2

    Pattern 3xx

    Replacement *

     

    pref 100

    trunk voip

    Pattern *

    Replacement *

     

    this is the log :

     

    [5] 2011/02/04 17:47:27:	SIP Rx udp:192.168.1.50:5060:
INVITE sip:348xxxxxxx@172.16.10.210;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613
To: <sip:348xxxxxxx@172.16.10.210;user=phone>
Call-ID: 240994b8@pbx
CSeq: 7987 INVITE
Max-Forwards: 70
Contact: <sip:123@192.168.1.50:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
P-Asserted-Identity: "vm" <sip:123@172.16.10.210>
Content-Type: application/sdp
Content-Length: 327

v=0
o=- 14816 14816 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 50782 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2011/02/04 17:47:27:	Identify trunk (IP address/port and domain match) 12
[5] 2011/02/04 17:47:27:	SIP Rx udp:192.168.1.50:5060:
INVITE sip:348xxxxxxx@172.16.10.210;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613
To: <sip:348xxxxxxx@172.16.10.210;user=phone>
Call-ID: 240994b8@pbx
CSeq: 7987 INVITE
Max-Forwards: 70
Contact: <sip:123@192.168.1.50:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
P-Asserted-Identity: "vm" <sip:123@172.16.10.210>
Content-Type: application/sdp
Content-Length: 327

v=0
o=- 14816 14816 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 50782 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2011/02/04 17:47:27:	SIP Tx udp:192.168.1.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport=5060
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613
To: <sip:348xxxxxxx@172.16.10.210;user=phone>;tag=d661df1270
Call-ID: 240994b8@pbx
CSeq: 7987 INVITE
Content-Length: 0

[5] 2011/02/04 17:47:27:	Domain trunk pm@172.16.10.210 could not identify user for 348xxxxxxx
[5] 2011/02/04 17:47:27:	SIP Tx udp:192.168.1.50:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport=5060
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613
To: <sip:348xxxxxxx@172.16.10.210;user=phone>;tag=d661df1270
Call-ID: 240994b8@pbx
CSeq: 7987 INVITE
Contact: <sip:123@172.16.10.210:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Length: 0

[5] 2011/02/04 17:47:27:	SIP Rx udp:192.168.1.50:5060:
ACK sip:348xxxxxxx@172.16.10.210;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613
To: <sip:348xxxxxxx@172.16.10.210;user=phone>;tag=d661df1270
Call-ID: 240994b8@pbx
CSeq: 7987 ACK
Max-Forwards: 70
Contact: <sip:123@192.168.1.50:5060;transport=udp>
P-Asserted-Identity: "vm" <sip:123@172.16.10.210>
Content-Length: 0

     

    i can configure the sla or the blf of the remote extension ??

  17. connect remote office through vpn

    in General Setup

    Posted

    i would connect two offices through a vpn connection, but I have many problems

     

    i have create two trunk gateway in this way:

     

    office1 ( 172.16.10.210 )

    Name: office2

    Type: sip gateway

    Direction: in and out

    Trunk Destination: generic sip server

    State: enabled

    Account: 123

    Domain: 192.168.1.60

    Username: 123

    Password: ****

    Proxy Address: 192.168.1.60

     

    office2 ( 192.168.1.60 )

    Name: office1

    Type: sip gateway

    Direction: in and out

    Trunk Destination: generic sip server

    State: enabled

    Account: 123

    Domain: 172.16.10.210

    Username: 123

    Password: ****

    Proxy Address: 172.16.10.210

     

    the extension in the office1 is 2xx and in the office2 is 3xx

     

    the dial plan for office1 is :

    pref 100

    Trunk office2

    Pattern: 3xx

    Replacement: *

     

    the dial plan for office2 is :

    pref 100

    Trunk office1

    Pattern: 2xx

    Replacement: *

     

    when i make a call from office1 to office2 and viceversa i give this error :

    [5] 2011/02/04 11:35:48:	SIP Rx udp:192.168.1.50:5060:
INVITE sip:200@172.16.10.210;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523
To: <sip:200@172.16.10.210;user=phone>
Call-ID: 9e1eef98@pbx
CSeq: 12399 INVITE
Max-Forwards: 70
Contact: <sip:123@192.168.1.50:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
P-Asserted-Identity: "vm" <sip:123@172.16.10.210>
Content-Type: application/sdp
Content-Length: 327

v=0
o=- 17786 17786 IN IP4 192.168.1.50
s=-
c=IN IP4 192.168.1.50
t=0 0
m=audio 55380 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

[5] 2011/02/04 11:35:48:	Last message repeated 2 times
[5] 2011/02/04 11:35:48:	SIP Tx udp:192.168.1.50:5060:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport=5060
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523
To: <sip:200@172.16.10.210;user=phone>;tag=d2057dfc98
Call-ID: 9e1eef98@pbx
CSeq: 12399 INVITE
Content-Length: 0

[5] 2011/02/04 11:35:48:	Received incoming call without trunk information and user has not been found
[5] 2011/02/04 11:35:48:	SIP Tx udp:192.168.1.50:5060:

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport=5060
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523
To: <sip:200@172.16.10.210;user=phone>;tag=d2057dfc98
Call-ID: 9e1eef98@pbx
CSeq: 12399 INVITE
Contact: <sip:200@172.16.10.210:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Length: 0

[5] 2011/02/04 11:35:48:	SIP Rx udp:192.168.1.50:5060:
ACK sip:200@172.16.10.210;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport
From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523
To: <sip:200@172.16.10.210;user=phone>;tag=d2057dfc98
Call-ID: 9e1eef98@pbx
CSeq: 12399 ACK
Max-Forwards: 70
Contact: <sip:123@192.168.1.50:5060;transport=udp>
P-Asserted-Identity: "vm" <sip:123@172.16.10.210>
Content-Length: 0

     

    i have not found any document about connect two office through a vpn connection

     

    thanks

  18. forwarding to Cell Phone on a different stages

    in Using Cell Phones

    Posted

    Well 100 Trying is typically not the last result you will see coming from the gateway. Search for other messages with the same "Call-ID". If you see a code between 180 and 199 then you will hear ringback (which is good), if you see a 200 code that means the call got connected (not good if is comes right after the INVITE request was sent out). VoIP providers are usually better as they dont have to run the calls through analog lines, which makes it a lot easier to find out if the call got connected.

     

    That is actually a killer reason IMHO for SIP trunking. You know when the call actually got connected...

     

    this is the complete message sip when i try to call from a ext to a hunt gruop with two different cell phone setting in the first e second stage :

     

    the isdn patton smart node 4554 is configured without autentication with the snom one, may depend from this ?

     

    [7] 2011/01/29 12:00:01:	SIP Rx tls:172.16.10.37:2070:
INVITE sip:71@172.16.10.201;user=phone SIP/2.0
Via: SIP/2.0/TLS 172.16.10.37:2070;branch=z9hG4bK-pz055tq2tqgw;rport
From: "Int 20" <sip:20@172.16.10.201>;tag=hnoxz8pdmi
To: <sip:71@172.16.10.201;user=phone>
Call-ID: 3c3486a9556c-w3p5he6qu0xz
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:20@172.16.10.37:2070;transport=tls;line=2wbbgcc9>;reg-id=1
X-Serialnumber: 00041331A667
P-Key-Flags: keys="3"
User-Agent: snom320/8.4.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 524

v=0
o=root 1254679951 1254679951 IN IP4 172.16.10.37
s=call
c=IN IP4 172.16.10.37
t=0 0
m=audio 57660 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:edjvoHy8AcFJNH2yvJpIQdW00KW3sVMc+/X2tupc
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv

[7] 2011/01/29 12:00:01:	SIP Tx tls:172.16.10.37:2070:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 172.16.10.37:2070;branch=z9hG4bK-pz055tq2tqgw;rport=2070
From: "Int 20" <sip:20@172.16.10.201>;tag=hnoxz8pdmi
To: <sip:71@172.16.10.201;user=phone>;tag=34bf619e2a
Call-ID: 3c3486a9556c-w3p5he6qu0xz
CSeq: 1 INVITE
Content-Length: 0

[7] 2011/01/29 12:00:01:	SIP Tx tls:172.16.10.37:2070:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 172.16.10.37:2070;branch=z9hG4bK-pz055tq2tqgw;rport=2070
From: "Int 20" <sip:20@172.16.10.201>;tag=hnoxz8pdmi
To: <sip:71@172.16.10.201;user=phone>;tag=34bf619e2a
Call-ID: 3c3486a9556c-w3p5he6qu0xz
CSeq: 1 INVITE
Contact: <sip:20@172.16.10.200:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3974
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 429

v=0
o=- 1449931 1449931 IN IP4 172.16.10.201
s=-
c=IN IP4 172.16.10.201
t=0 0
m=audio 58452 RTP/AVP 0 8 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:MeNKoVukSlSggKuXWOzmhNLuc8u1gfcTz80bJY8L
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

[7] 2011/01/29 12:00:01:	SIP Tx udp:172.16.10.205:5060:
INVITE sip:348xxxxxxx@172.16.10.205:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK-a433dfa59d493e40369041e8a83b99e7;rport
From: "Int 20" <sip:20@172.16.10.201;user=phone>;tag=356689874
To: <sip:348xxxxxxx@172.16.10.205:5060;user=phone>
Call-ID: 19ebdeb8@pbx
CSeq: 29855 INVITE
Max-Forwards: 70
Contact: <sip:039xxxxxxxx@172.16.10.200:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3974
P-Asserted-Identity: "Isdn" <sip:039xxxxxxxx@172.16.10.205:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=- 1949439694 1949439694 IN IP4 172.16.10.201
s=-
c=IN IP4 172.16.10.201
t=0 0
m=audio 56242 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=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[7] 2011/01/29 12:00:01:	SIP Rx tls:172.16.10.37:2070:
PRACK sip:20@172.16.10.200:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.16.10.37:2070;branch=z9hG4bK-zzk68e8yqt22;rport
From: "Int 20" <sip:20@172.16.10.201>;tag=hnoxz8pdmi
To: <sip:71@172.16.10.201;user=phone>;tag=34bf619e2a
Call-ID: 3c3486a9556c-w3p5he6qu0xz
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:20@172.16.10.37:2070;transport=tls;line=2wbbgcc9>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons
Content-Length: 0

[7] 2011/01/29 12:00:01:	SIP Tx tls:172.16.10.37:2070:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 172.16.10.37:2070;branch=z9hG4bK-zzk68e8yqt22;rport=2070
From: "Int 20" <sip:20@172.16.10.201>;tag=hnoxz8pdmi
To: <sip:71@172.16.10.201;user=phone>;tag=34bf619e2a
Call-ID: 3c3486a9556c-w3p5he6qu0xz
CSeq: 2 PRACK
Contact: <sip:20@172.16.10.200:5061;transport=tls>
User-Agent: snom-PBX/2011-4.2.0.3974
Content-Length: 0

[7] 2011/01/29 12:00:01:	SIP Rx udp:172.16.10.205:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK-a433dfa59d493e40369041e8a83b99e7;rport=5060;received=172.16.10.200
From: "Int 20" <sip:20@172.16.10.201;user=phone>;tag=356689874
To: <sip:348xxxxxxx@172.16.10.205:5060;user=phone>
Call-ID: 19ebdeb8@pbx
CSeq: 29855 INVITE
Server: Patton SN4554 2BIS EUI 00A0BA05EA30 R5.5 2010-09-03 SIP M5T SIP Stack/4.0.28.28
Content-Length: 0

[7] 2011/01/29 12:00:05:	SIP Rx udp:172.16.10.205:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK-a433dfa59d493e40369041e8a83b99e7;rport=5060;received=172.16.10.200
From: "Int 20" <sip:20@172.16.10.201;user=phone>;tag=356689874
To: <sip:348xxxxxxx@172.16.10.205:5060;user=phone>;tag=1624736971
Call-ID: 19ebdeb8@pbx
CSeq: 29855 INVITE
Contact: <sip:348xxxxxxx@172.16.10.205:5060>
Server: Patton SN4554 2BIS EUI 00A0BA05EA30 R5.5 2010-09-03 SIP M5T SIP Stack/4.0.28.28
Content-Length: 0

[7] 2011/01/29 12:00:08:	SIP Rx udp:172.16.10.205:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK-a433dfa59d493e40369041e8a83b99e7;rport=5060;received=172.16.10.200
From: "Int 20" <sip:20@172.16.10.201;user=phone>;tag=356689874
To: <sip:348xxxxxxx@172.16.10.205:5060;user=phone>;tag=1624736971
Call-ID: 19ebdeb8@pbx
CSeq: 29855 INVITE
Contact: <sip:348xxxxxxx@172.16.10.205:5060>
Server: Patton SN4554 2BIS EUI 00A0BA05EA30 R5.5 2010-09-03 SIP M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 43 IN IP4 172.16.10.205
s=SIP Call
c=IN IP4 172.16.10.205
t=0 0
m=audio 4948 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=sendrecv
[7] 2011/01/29 12:00:08:	Call 19ebdeb8@pbx: Clear last INVITE
[7] 2011/01/29 12:00:08:	SIP Tx udp:172.16.10.205:5060:
ACK sip:348xxxxxxx@172.16.10.205:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.200:5060;branch=z9hG4bK-6fdeb9310a1a93e0390932bb5b0ca802;rport
From: "Int 20" <sip:20@172.16.10.201;user=phone>;tag=356689874
To: <sip:348xxxxxxx@172.16.10.205:5060;user=phone>;tag=1624736971
Call-ID: 19ebdeb8@pbx
CSeq: 29855 ACK
Max-Forwards: 70
Contact: <sip:039xxxxxxxx@172.16.10.200:5060;transport=udp>
P-Asserted-Identity: "Isdn" <sip:039xxxxxxxx@172.16.10.205:5060>
Content-Length: 0

[7] 2011/01/29 12:00:08:	SIP Tx tls:172.16.10.37:2070:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 172.16.10.37:2070;branch=z9hG4bK-pz055tq2tqgw;rport=2070
From: "Int 20" <sip:20@172.16.10.201>;tag=hnoxz8pdmi
To: <sip:71@172.16.10.201;user=phone>;tag=34bf619e2a
Call-ID: 3c3486a9556c-w3p5he6qu0xz
CSeq: 1 INVITE
Contact: <sip:20@172.16.10.200:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3974
Content-Type: application/sdp
Content-Length: 429

v=0
o=- 1449931 1449931 IN IP4 172.16.10.201
s=-
c=IN IP4 172.16.10.201
t=0 0
m=audio 58452 RTP/AVP 0 8 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:MeNKoVukSlSggKuXWOzmhNLuc8u1gfcTz80bJY8L
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

[7] 2011/01/29 12:00:09:	SIP Rx tls:172.16.10.37:2070:
ACK sip:20@172.16.10.200:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.16.10.37:2070;branch=z9hG4bK-lofu8rngdav2;rport
From: "Int 20" <sip:20@172.16.10.201>;tag=hnoxz8pdmi
To: <sip:71@172.16.10.201;user=phone>;tag=34bf619e2a
Call-ID: 3c3486a9556c-w3p5he6qu0xz
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:20@172.16.10.37:2070;transport=tls;line=2wbbgcc9>;reg-id=1
Proxy-Require: buttons
Content-Length: 0

  19. forwarding to Cell Phone on a different stages

    in Using Cell Phones

    Posted

    Well the problem could be that the call connects as soon as you dial the number, then the hunt group would obviously stop. For example, when calling the cell phone the mailbox could pick up. Terminating traffic in the analog world is not so wasy and many gateways send the connected signal immediatly (you will still just hear ringback tone); that's because it is so hard to figure out if the call is connected or not in the analog world.

     

    You can check in the SIP messages if the gateway sends a 200 Ok response on the INVITE request.

     

    in fact ... after the invite message there is not the "OK" but the there is the "100 trying" message

    So the only solution is to use a voip provider ?

  20. forwarding to Cell Phone on a different stages

    in Using Cell Phones

    Posted

    What version are you on? Does it work with no cell phones involved (just regular extensions with no cell phone forwarding)?

     

    hi,

     

    my version is 4.2.0.3974 on centos 32bit, if i insert the extensions in the stages there is not problem, also if i use the my voip provider there is not problem...

    the issue appears when i use the isdn line connected with the snom one through the patton 4554 !!! is a configuration problem or a problem of telecom ?

  21. forwarding to Cell Phone on a different stages

    in Using Cell Phones

    Posted

    i need to forwarding to Cell Phone on a different stages ...

    i explain ... stage 1 call the cell phone 123456789 after 20 sec if not responding go to the stage 2 and call the cell phone 987654321 after 20 sec go to the stage 3 ecc...

     

    i tried to make it with the hunt group but the forward does not go over the first stage ... ring only the first cell phone ..

     

    i tried to insert the extension in the stage of the hunt group rather the number of the cell phone and i have enabled the "When calling the extension in a hunt group" under redirection parameters but nothing change ... only the first cell phone ring

     

    it is a bug of the snom one ???

     

    someone can give me some advice please ?

     

    thanks

  22. Multicast pnp with snom m9

    in snom Phones

    Posted

    You can provision also multiple handsets. For this you can assign the MAC into multiple extensions.

     

    If the multicast PnP should not work, you can use the IP address of the PBX and put it into the Provisioning Server setting (network section in the web interface of the m9). If that does not work you probably have a problem with the certificates (what does the PBX say in the certificates who it trusts?).

     

    the phone m9 does not have a mac address this are identified by a handset ID, only the the base have one mac, i must assign the same mac into the multiple extensions ?

  23. Multicast pnp with snom m9

    in snom Phones

    Posted

    i try to connect a snom 320 for exclude problem with multicast

    another thing that i not understand, the m9 kit have 2 phone, the provision configure the first identity of the snom, and for configure the other ?

    i can configure it with the provision or i must configure it manualy ?

  24. incoming call on the fax numbers

    in General Setup

    Posted

    In my case, unfortunately, fax machines are outside the pbx and are not connected to an ATA but direcly connected to the isdn line.

    The only solution that i found is modify the patton 4554 so that it not forwards to the snom one the calls direct to the fax numbers.

     

    On the patton 4554 in the "call-router" under the "routing table" i have added this entry for each trunk with the pbx :

    called-e164 : my-fax-number Destination: none

     

    thanks

  25. Multicast pnp with snom m9

    in snom Phones

    Posted

    hi,

    i have a snom m9 kit, with one base a two phone, i tried to configure it with the multicast pnp, i have inserted the mac in the "Bind to MAC Address" filed under the profile account, i restarted the m9 but nothing happens, the phone don't does the provisioning.

×
×
  • Create New...