pbxuser911

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

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

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?

sorry for not being so clear here is what i want, if the caller is in the Queue for more then 45 seconds, it should get transferred to another extension, not ring that extension, but get transferred to that extension

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

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

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.

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?

where you add that 1 should be for sales add 1# the # will add a delay, so if you type 123 it will ring extension 123 and not think its option 1 for sales

how do i upload the image? thats all i got from them they wouldnt send me a text log or anything like that

do they make any devices that can be used just for intercom? maybe something like a VOIP Paging device?

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?

Has anyone on here used Enable 911? they claim when they try to call us back (during the test process) they keep getting a 404 error any help?

Has anyone used Sipp with PCAP PLAY support to stress test thier system and had good luck getting true results?

would also be nice to have #411 for the company directory

is there a file that sounds better? that sound sounded wierd for our customers, they were asking us what is that wierd sound so we removed it

when saying yes to Say "Please enter the extension number" it will only say Say "Please enter the extension number" it wont say "this extension does not exist" it would be nice to have the optiont o tell the system what to do, either go back one step, or tell it what extension to go to, or return to the Auto Attendant etc

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?

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

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 :

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

where is the log keep setting located?

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?

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

is it possible to maybe program a key on the phone to do that feuture? if not then the grandstream phones have one over the linksys as far this is option