lirees

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

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

Hi, I can manage the service flag with a call from the outside? I'll explain .... if i forget to activate the service flag i would like to turn on even when i'm outside of my office whit a call form my cell phone. is it possible? thanks

Yesss !!! Thanks

the version is 2011-4.2.1.4025

Hi, how can I customize the template with the snom one soho ? in the web interface there is not the tab "web page controll"

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 ???

opsss i have posted in the AA section rather the IVR .... sorry

hello to everybody there is any way to listen the message recorded with the dial code from phone ??? i usualy use the IVR with the service flag for customize a message when the office is closed, but i can only record the message, for hear the message i have to enable the SF and call the office thanks

hi, how can i do to block an specific extension for the outgoing call ??

there's a way to put the call on hold through the dial code ( not the call parking ) ???

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

great ... it worked !!! thanks

great !!! this is a fantastic workaround thank so much but

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

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 ??

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

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

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 ?

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 ?

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

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 ?

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 ?

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

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.