pbxuser911

I need help with setting up the following failover solution i have 3 trunk's and each of those 3 have a 2nd trunk group which the providor gave us to use for failover in the main trunk i have Failover Behavior on all error codes, request timeout 3 here is how i have the dial plan set up: PREF 100 - Unassigned PREF 101 - E911 SIPGW 1 PREF 102 - E911 SIPGW 2 PREF 104 VOICE TRADING SIPGW 1 PATTERN 011* REPLACEMENT 00* PREF 105 VOICE TRADING SIPGW 2 PATTERN 011* REPLACEMENT 00* PREF 107 VP SIPGW 1 PATTERN 1800* REPLACEMENT 1800* PREF 107 VP SIPGW 1 PATTERN 1877* REPLACEMENT 1877* PREF 108 VP SIPGW 2 PATTERN 1800* REPLACEMENT 1800* PREF 108 VP SIPGW 2 PATTERN 1877* REPLACEMENT 1877* Now the way this is set up, if the trunk on PREF 107 VP SIPGW 1 is down, will it failover to PREF 108 VP SIPGW 2? and what should i do if i want to add another trunk (a diff termination) in case V SIPGW1 & VP SIPGW2 goes down? do i add lets say TSG SIPGW1 trunk and set PREF 109 TSG SIPGW1 PATTERN 1800* REPLACEMENT 1800* ? and PREF 109 TSG SIPGW2 PATTERN 1877* REPLACEMENT 1877* ? and do i need to enable failover on the trunk VP SIPGW2 on all error codes?

any help here? maybe someone tell me how to i set up that when i dial a 10 digit number it will automaticly add a 1 in frotn of the number?

here is what my auto attendtant account number(s) look like "010 (972)555-1212 (972)555-1313 (972)555-1414" without the "

I’m glad we are not the only one our system keep crashing almost every day due to that problem a new release was given to us by the programmers, which that actually caused our system to lose all the configuration files so unless they come out with a newer release then 3.2 i wouldn’t suggest updating it on the other note the only way to fix this, I’m thinking is to do the following disable the login page to come up when entering the IP address of the server, and only allow it for one IP address, which will be the main office that provisions the phones and creates the domains etc this way no subscriber has access to it. anyone that can come up with a fix, please advise BTW what OS are you running? at first we were told it keeps crashing because we are using Fedora, but now even with FreeBSD it also keep crashing if you log in with the invalid credentials (which customers keep trying since they forget their password,) wouldn’t it be nice to have a "forgot my password" tool? and if we do have it, allow us to Block web log in per extension, maybe one company doesn’t want their employee logging on to the web portal to change his email address etc.... and if we have a forgot password, he will get his password which isn’t good, so having an option if disabling web log in per extension would be awesome

does anyone have any step by step instructions on how to change the email thats being sent out?

So how do i change the format of the email thats beign sent out when i get a missed call or a voice mail etc?

what did you do to get it working?

if i have a country code of 1 then the caller ID looks like this when someone calls us +1xxxxxxxxxx how do i remove the +?

is there anyway to set it up in the dialplan that when you dial a 10 digit number it should add the number 1 in front of it? im also havignt he same problem that when i dial from the missed call log, it doesnt have the 1 in front of it or maybe can we have PBXnSIP add a number 1 in front of the number when display the caller ID? it should only add it on USA 10 digit numbers (so it wont add it on international calls)

has anyone figured out how to use multicast with this phone?

Is there any way to restore a pbxnsip configuration from backup without having to restart the pbxnsip service? the problem we're having is that we're doing some simple failover but when we rsync the config files found in this script: tar cvfz $filename acds adrbook attendants autocallback button_lists buttons callingcards cdr colines conferences dial_plan dial_plan_entry domain_alias domains extensions generated hoots hunts ivrnodes lamps messages mohs orbits pnp_parms recordings regidx registrations schedules shared_lines spool srvflags tftp trunks user_alias users pbx.xml we have to restart pbxnsip on the second server for the changes to come up.

How do you share a voice mail box? i want to have a general voice mail that every employee here inthe office will know when there is a message left there, and should also be able to receive its own voice mails i have 3 general voice mail boxes at least which i want to be able to share

We would like to have the following done, can anyone advise in plain english the best way to do it ? I want when someone calls into our customer service que, it should ring my cell phone and also my other phone number (home office) should also ring in additionan when i answer my cell phone or home office phone it should have the option of pressing 1 to accept or 2 to ignore or 3 to do some other cool things, maybe transfer it to another 10 digit phone number, and if i ignore it then it should keep ringing on the other extensions that are on that que,

Regarding paging is there a way to give a busy tone if someone pages a phone that is already on the line with someone else. It seems like a poor decision that Linksys made to have a page automatically put a call on hold. Say your on the phone with a customer and another extension pages you, your customer randomly gets put on hold without them even knowing and the other extension has no other way of knowing you were on. I hope Linksys does something about these issue ASAP because it really was a poor move to sell a product advertising it to do certain features that it currently does not support.

whats if we dont want to use cyberdata or any other added devices, just simply use the phone itsself? has anyone configured a SPA962 phone to work?

yes and nothign happends... the call will not redirect to the extension entered in there it still rings all extensions even after the 20 seconds or whatever i set as the time

i am using 3.1.1.3110

When i go to the Currently Active Calls i see a blank call which says it started at: 2106/02/06 01:28:16 and it doesnt have info for "from" "to" "state" but gives me the option to X the call is this something someone has ever seen?

i told TSG what you recomended me telling them about the RFC, Looks like the "TSG_Global_GW" has a little issue with RFC3261: CODE quoted-string = SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE qdtext = LWS / %x21 / %x23-5B / %x5D-7E / UTF8-NONASCII Unfortunately, the PBX just follows the RFC rules; there is no way to perform a special escaping of characters. The best solution if the gateway gets a software update with a fix. and this is what he replyed back to me---- I will look into the call issue, however in terms of the RFC it states the following note the “ “ around the name, however when we get calls from your switch the “ “ are missing and thus when a charter other then a-z is used it causes the system to break. Link for Ref http://www.ietf.org/rfc/rfc3261.txt 20.20 From The From header field indicates the initiator of the request. This may be different from the initiator of the dialog. Requests sent by the callee to the caller use the callee's address in the From header field. The optional "display-name" is meant to be rendered by a human user interface. A system SHOULD use the display name "Anonymous" if the identity of the client is to remain hidden. Even if the "display- name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, question mark, or semicolon. Syntax issues are discussed in Section 7.3.1. Two From header fields are equivalent if their URIs match, and their parameters match. Extension parameters in one header field, not present in the other are ignored for the purposes of comparison. This means that the display name and presence or absence of angle brackets do not affect matching. See Section 20.10 for the rules for parsing a display name, URI and URI parameters, and header field parameters. The compact form of the From header field is f. Examples: From: "A. G. Bell" <sip:agb@bell-telephone.com> ;tag=a48s From: sip:+12125551212@server.phone2net.com;tag=887s f: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8 20.10 Contact A Contact header field value provides a URI whose meaning depends on the type of request or response it is in. A Contact header field value can contain a display name, a URI with URI parameters, and header parameters. This document defines the Contact parameters "q" and "expires". These parameters are only used when the Contact is present in a REGISTER request or response, or in a 3xx response. Additional parameters may be defined in other specifications. When the header field value contains a display name, the URI including all URI parameters is enclosed in "<" and ">". If no "<" and ">" are present, all parameters after the URI are header parameters, not URI parameters. The display name can be tokens, or a quoted string, if a larger character set is desired. Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon, or question mark. There may or may not be LWS between the display-name and the "<". These rules for parsing a display name, URI and URI parameters, and header parameters also apply for the header fields To and From. The Contact header field has a role similar to the Location header field in HTTP. However, the HTTP header field only allows one address, unquoted. Since URIs can contain commas and semicolons as reserved characters, they can be mistaken for header or parameter delimiters, respectively. The compact form of the Contact header field is m (for "moved"). Examples: Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com> ;q=0.7; expires=3600, "Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1 m: <sips:bob@192.0.2.4>;expires=60

THIS IS ME CALLING THEM WITHOUT THE PREFIX OF 1 ON THAT TRUNK------------ [3] 2009/01/09 11:37:05: Could not open WAV file audio_moh/noise.wav [3] 2009/01/09 11:37:05: Last message repeated 7 times [5] 2009/01/09 11:37:05: SIP Rx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: INVITE sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK0730087281b33c03 From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM> Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: replaces, timer, path Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36154 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 406 v=0 o=2486 8000 8000 IN IP4 192.168.2.160 s=SIP Call c=IN IP4 192.168.2.160 t=0 0 m=audio 5092 RTP/AVP 0 8 18 4 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 [5] 2009/01/09 11:37:05: Identify trunk (domain name match) 39 [5] 2009/01/09 11:37:05: SIP Tx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK0730087281b33c03;rport=41365;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX From: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36154 INVITE Content-Length: 0 [5] 2009/01/09 11:37:05: SIP Tx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK0730087281b33c03;rport=41365;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36154 INVITE User-Agent: ----/3.1.1.3110 WWW-Authenticate: Digest realm="COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM",nonce="84742eb46686fb7e690d86bec8b9381b",domain="sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM",algorithm=MD5 Content-Length: 0 [3] 2009/01/09 11:37:05: Could not open WAV file audio_moh/noise.wav [3] 2009/01/09 11:37:05: Last message repeated 2 times [5] 2009/01/09 11:37:05: SIP Rx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: ACK sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK0730087281b33c03 From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: path Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36154 ACK User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 [3] 2009/01/09 11:37:05: Could not open WAV file audio_moh/noise.wav [5] 2009/01/09 11:37:05: SIP Rx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: INVITE sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK328bcb5b613c47eb From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM> Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: replaces, timer, path Authorization: Digest username="2486", realm="COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM", algorithm=MD5, uri="sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM", nonce="84742eb46686fb7e690d86bec8b9381b", response="dd3d398922646124171219c1334aea69" Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36155 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 406 v=0 o=2486 8000 8001 IN IP4 192.168.2.160 s=SIP Call c=IN IP4 192.168.2.160 t=0 0 m=audio 5092 RTP/AVP 0 8 18 4 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 [5] 2009/01/09 11:37:05: SIP Tx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK328bcb5b613c47eb;rport=41365;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36155 INVITE Content-Length: 0 [5] 2009/01/09 11:37:05: SIP Tx udp:208.71.179.10:5060: INVITE sip:911@208.71.179.10;user=phone SIP/2.0 Via: SIP/2.0/UDP XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;branch=z9hG4bK-d11d73fb59b33e39c5005100ba9ff965;rport From: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=1641834112 To: <sip:911@208.71.179.10;user=phone> Call-ID: d8ff6a4f@pbx CSeq: 31628 INVITE Max-Forwards: 70 Contact: <sip:2486@XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: ----/3.1.1.3110 P-Asserted-Identity: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM> Content-Type: application/sdp Content-Length: 298 v=0 o=- 203250974 203250974 IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX s=- c=IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX t=0 0 m=audio 56660 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 [5] 2009/01/09 11:37:05: SIP Rx udp:208.71.179.138:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;branch=z9hG4bK-d11d73fb59b33e39c5005100ba9ff965;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX;rport=5060 From: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=1641834112 To: <sip:911@208.71.179.10;user=phone> Call-ID: d8ff6a4f@pbx CSeq: 31628 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> Content-Length: 0 [5] 2009/01/09 11:37:05: SIP Tx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK328bcb5b613c47eb;rport=41365;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36155 INVITE Contact: <sip:2486@XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: ----/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1270820695 1270820695 IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX s=- c=IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX t=0 0 m=audio 28142 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=ptime:20 a=sendrecv [5] 2009/01/09 11:37:05: SIP Rx udp:208.71.179.138:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;branch=z9hG4bK-d11d73fb59b33e39c5005100ba9ff965;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX;rport=5060 From: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=1641834112 To: <sip:911@208.71.179.10;user=phone>;tag=as0c12f027 Call-ID: d8ff6a4f@pbx CSeq: 31628 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> Content-Length: 0 [3] 2009/01/09 11:37:06: Could not open WAV file audio_moh/noise.wav [3] 2009/01/09 11:37:06: Last message repeated 15 times [5] 2009/01/09 11:37:06: SIP Tr udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK328bcb5b613c47eb;rport=41365;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36155 INVITE Contact: <sip:2486@XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: ----/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1270820695 1270820695 IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX s=- c=IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX t=0 0 m=audio 28142 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=ptime:20 a=sendrecv [5] 2009/01/09 11:37:06: SIP Rx udp:208.71.179.138:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;branch=z9hG4bK-d11d73fb59b33e39c5005100ba9ff965;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX;rport=5060 From: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=1641834112 To: <sip:911@208.71.179.10;user=phone>;tag=as0c12f027 Call-ID: d8ff6a4f@pbx CSeq: 31628 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> Content-Length: 0 [5] 2009/01/09 11:37:07: SIP Tr udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK328bcb5b613c47eb;rport=41365;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36155 INVITE Contact: <sip:2486@XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: ----/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1270820695 1270820695 IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX s=- c=IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX t=0 0 m=audio 28142 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=ptime:20 a=sendrecv [5] 2009/01/09 11:37:08: SIP Rx udp:208.71.179.138:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;branch=z9hG4bK-d11d73fb59b33e39c5005100ba9ff965;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX;rport=5060 From: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=1641834112 To: <sip:911@208.71.179.10;user=phone>;tag=as0c12f027 Call-ID: d8ff6a4f@pbx CSeq: 31628 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> 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 10004 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] 2009/01/09 11:37:08: SIP Tx udp:208.71.179.138:5060: ACK sip:911@208.71.179.138 SIP/2.0 Via: SIP/2.0/UDP XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;branch=z9hG4bK-c8cb44e3c04a96b83d49cf9a900256a3;rport From: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=1641834112 To: <sip:911@208.71.179.10;user=phone>;tag=as0c12f027 Call-ID: d8ff6a4f@pbx CSeq: 31628 ACK Max-Forwards: 70 Contact: <sip:2486@XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060;transport=udp> P-Asserted-Identity: "2486" <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM> Content-Length: 0 [5] 2009/01/09 11:37:08: SIP Tx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK328bcb5b613c47eb;rport=41365;received=XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36155 INVITE Contact: <sip:2486@XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: ----/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1270820695 1270820695 IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX s=- c=IN IP4 XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX t=0 0 m=audio 28142 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=ptime:20 a=sendrecv [5] 2009/01/09 11:37:08: SIP Rx udp:XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:41365: ACK sip:2486@XX.(IP ADDRESS WHERE THE PHONE IS LOCATED).XX:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK58bd874fda3fd108 From: "2486 " <sip:2486@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=bd4300c1d1a98b7c To: <sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM>;tag=dd5a7a0bb3 Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: path Authorization: Digest username="2486", realm="COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM", algorithm=MD5, uri="sip:911@COMPANY1.(DOMAIN POINTED TO PBXNSIP SERVER)OURCOMPANY.COM", nonce="84742eb46686fb7e690d86bec8b9381b", response="dd3d398922646124171219c1334aea69" Call-ID: 828c34a6ddc73d7b@192.168.2.160 CSeq: 36155 ACK User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 HERE IS THEM CALLING ME BACK------------ [5] 2009/01/09 11:38:22: SIP Rx udp:208.71.179.138:5060: INVITE sip:2486@XX.(SERVERS IP ADDRESS).XX SIP/2.0 Via: SIP/2.0/UDP 208.71.179.138:5060;branch=z9hG4bK69c0fa3b;rport From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as19e804c3 To: <sip:2486@XX.(SERVERS IP ADDRESS).XX> Contact: <sip:5147452143@208.71.179.138> Call-ID: 3f588d5e4ed06a425ead36417c5273a1@208.71.179.138 CSeq: 102 INVITE User-Agent: 911Enable SBC1 Max-Forwards: 70 Date: Fri, 09 Jan 2009 16:40:45 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 18710 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] 2009/01/09 11:38:22: SIP Tx udp:208.71.179.138:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 208.71.179.138:5060;branch=z9hG4bK69c0fa3b;rport=5060 From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as19e804c3 To: <sip:2486@XX.(SERVERS IP ADDRESS).XX>;tag=401db7e358 Call-ID: 3f588d5e4ed06a425ead36417c5273a1@208.71.179.138 CSeq: 102 INVITE Content-Length: 0 [5] 2009/01/09 11:38:22: 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=z9hG4bK69c0fa3b;rport=5060 From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as19e804c3 To: <sip:2486@XX.(SERVERS IP ADDRESS).XX>;tag=401db7e358 Call-ID: 3f588d5e4ed06a425ead36417c5273a1@208.71.179.138 CSeq: 102 INVITE Contact: <sip:2486@XX.(SERVERS IP ADDRESS).XX:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: -----/3.1.1.3110 Content-Length: 0 [5] 2009/01/09 11:38:22: Last message repeated 2 times [5] 2009/01/09 11:38:22: SIP Rx udp:208.71.179.138:5060: ACK sip:2486@XX.(SERVERS IP ADDRESS).XX SIP/2.0 Via: SIP/2.0/UDP 208.71.179.138:5060;branch=z9hG4bK69c0fa3b;rport From: "5147452143" <sip:5147452143@208.71.179.138>;tag=as19e804c3 To: <sip:2486@XX.(SERVERS IP ADDRESS).XX>;tag=401db7e358 Contact: <sip:5147452143@208.71.179.138> Call-ID: 3f588d5e4ed06a425ead36417c5273a1@208.71.179.138 CSeq: 102 ACK User-Agent: 911Enable SBC1 Max-Forwards: 70 Content-Length: 0

i added to log the trunk events, ill set up another test call with enable 911

what does the ! do?

you sure thats the auto attendant that answers your call from your cellphone?

i create a backup fromthe Save/Restore Configuration tool and the file is 0KB, does that make sense? i would expect it to be a little larger file, i dont want to try and use this file to restore cause then ill loose everything anyhelp would be appriciated

this is my call to enable 911, and thier system tryes calling us back... [5] 2009/01/08 14:18:39: SIP Rx udp:66.128.2.136:5060: OPTIONS sip:metaswitch@xx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKsa8c4e107oohoes986k1.1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info Max-Forwards: 69 Call-ID: E49F8B2B@sip10.xtele.com From: <sip:metaswitch@192.168.9.40:5060>;tag=sip10.xtele.com+1+0+470537ad CSeq: 165329127 OPTIONS Organization: MetaSwitch Supported: 100rel Content-Length: 0 Contact: <sip:metaswitch@66.128.2.136:5060;transport=udp> To: <sip:metaswitch@xx.xxx.xxx.xxx> [5] 2009/01/08 14:18:39: SIP Tx udp:66.128.2.136:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 66.128.2.136:5060;branch=z9hG4bKsa8c4e107oohoes986k1.1 From: <sip:metaswitch@192.168.9.40:5060>;tag=sip10.xtele.com+1+0+470537ad To: <sip:metaswitch@xx.xxx.xxx.xxx>;tag=bf5a15a9d9 Call-ID: E49F8B2B@sip10.xtele.com CSeq: 165329127 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [5] 2009/01/08 14:18:58: SIP Rx udp:xx.xx.xxx.xx:41365: INVITE sip:911@our.company.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK9d3cbb85fe454727 From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com> Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: replaces, timer, path Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58005 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 406 v=0 o=2486 8000 8000 IN IP4 192.168.2.160 s=SIP Call c=IN IP4 192.168.2.160 t=0 0 m=audio 5066 RTP/AVP 0 8 18 4 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 [5] 2009/01/08 14:18:58: SIP Tx udp:xx.xx.xxx.xx:41365: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK9d3cbb85fe454727;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58005 INVITE Content-Length: 0 [5] 2009/01/08 14:18:58: SIP Tx udp:xx.xx.xxx.xx:41365: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK9d3cbb85fe454727;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58005 INVITE User-Agent: MIKE/3.1.1.3110 WWW-Authenticate: Digest realm="our.company.com",nonce="9187416fad0b35ed973e718a1e137b40",domain="sip:911@our.company.com",algorithm=MD5 Content-Length: 0 [5] 2009/01/08 14:18:58: SIP Rx udp:xx.xx.xxx.xx:41365: ACK sip:911@our.company.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK9d3cbb85fe454727 From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: path Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58005 ACK User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 [5] 2009/01/08 14:18:58: SIP Rx udp:xx.xx.xxx.xx:41365: INVITE sip:911@our.company.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK01bfde22a7e456d4 From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com> Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: replaces, timer, path Authorization: Digest username="2486", realm="our.company.com", algorithm=MD5, uri="sip:911@our.company.com", nonce="9187416fad0b35ed973e718a1e137b40", response="4aefc8322bfd5e97ccd41583d0524294" Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58006 INVITE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 406 v=0 o=2486 8000 8001 IN IP4 192.168.2.160 s=SIP Call c=IN IP4 192.168.2.160 t=0 0 m=audio 5066 RTP/AVP 0 8 18 4 2 97 9 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 [5] 2009/01/08 14:18:58: SIP Tx udp:xx.xx.xxx.xx:41365: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK01bfde22a7e456d4;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58006 INVITE Content-Length: 0 [5] 2009/01/08 14:18:58: SIP Tx udp:208.71.179.10:5060: INVITE sip:911@208.71.179.10;user=phone SIP/2.0 Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-3db0bd43de88e63a5f05ffc2f31ea103;rport From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone> Call-ID: dc81ee01@pbx CSeq: 27504 INVITE Max-Forwards: 70 Contact: <sip:12486@xx.xxx.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: MIKE/3.1.1.3110 P-Asserted-Identity: "2486 2486" <sip:12486@our.company.com;user=phone> Content-Type: application/sdp Content-Length: 300 v=0 o=- 1202781086 1202781086 IN IP4 xx.xxx.xxx.xxx s=- c=IN IP4 xx.xxx.xxx.xxx t=0 0 m=audio 17420 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 [5] 2009/01/08 14:18:58: SIP Rx udp:208.71.179.138:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-3db0bd43de88e63a5f05ffc2f31ea103;received=xx.xxx.xxx.xxx;rport=5060 From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone> Call-ID: dc81ee01@pbx CSeq: 27504 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> Content-Length: 0 [3] 2009/01/08 14:18:58: Could not open WAV file audio_moh/noise.wav [5] 2009/01/08 14:18:58: SIP Tx udp:xx.xx.xxx.xx:41365: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK01bfde22a7e456d4;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58006 INVITE Contact: <sip:2486@xx.xxx.xxx.xxx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: MIKE/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1172077554 1172077554 IN IP4 xx.xxx.xxx.xxx s=- c=IN IP4 xx.xxx.xxx.xxx t=0 0 m=audio 25236 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=ptime:20 a=sendrecv [3] 2009/01/08 14:18:58: Could not open WAV file audio_moh/noise.wav [3] 2009/01/08 14:18:58: Last message repeated 2 times [5] 2009/01/08 14:18:58: SIP Rx udp:208.71.179.138:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-3db0bd43de88e63a5f05ffc2f31ea103;received=xx.xxx.xxx.xxx;rport=5060 From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone>;tag=as5af89d5b Call-ID: dc81ee01@pbx CSeq: 27504 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> Content-Length: 0 [5] 2009/01/08 14:18:59: SIP Tr udp:xx.xx.xxx.xx:41365: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK01bfde22a7e456d4;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58006 INVITE Contact: <sip:2486@xx.xxx.xxx.xxx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: MIKE/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1172077554 1172077554 IN IP4 xx.xxx.xxx.xxx s=- c=IN IP4 xx.xxx.xxx.xxx t=0 0 m=audio 25236 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=ptime:20 a=sendrecv [5] 2009/01/08 14:18:59: SIP Rx udp:208.71.179.138:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-3db0bd43de88e63a5f05ffc2f31ea103;received=xx.xxx.xxx.xxx;rport=5060 From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone>;tag=as5af89d5b Call-ID: dc81ee01@pbx CSeq: 27504 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> Content-Length: 0 [5] 2009/01/08 14:19:00: SIP Tr udp:xx.xx.xxx.xx:41365: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK01bfde22a7e456d4;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58006 INVITE Contact: <sip:2486@xx.xxx.xxx.xxx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: MIKE/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1172077554 1172077554 IN IP4 xx.xxx.xxx.xxx s=- c=IN IP4 xx.xxx.xxx.xxx t=0 0 m=audio 25236 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=ptime:20 a=sendrecv [5] 2009/01/08 14:19:02: SIP Rx udp:208.71.179.138:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-3db0bd43de88e63a5f05ffc2f31ea103;received=xx.xxx.xxx.xxx;rport=5060 From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone>;tag=as5af89d5b Call-ID: dc81ee01@pbx CSeq: 27504 INVITE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> 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 16626 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 14:19:02: SIP Tx udp:208.71.179.138:5060: ACK sip:911@208.71.179.138 SIP/2.0 Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-0181ff85dcdfba9a576b17d4e9524947;rport From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone>;tag=as5af89d5b Call-ID: dc81ee01@pbx CSeq: 27504 ACK Max-Forwards: 70 Contact: <sip:12486@xx.xxx.xxx.xxx:5060;transport=udp> P-Asserted-Identity: "2486 2486" <sip:12486@our.company.com;user=phone> Content-Length: 0 [5] 2009/01/08 14:19:02: SIP Tx udp:xx.xx.xxx.xx:41365: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK01bfde22a7e456d4;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58006 INVITE Contact: <sip:2486@xx.xxx.xxx.xxx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: MIKE/3.1.1.3110 Content-Type: application/sdp Content-Length: 312 v=0 o=- 1172077554 1172077554 IN IP4 xx.xxx.xxx.xxx s=- c=IN IP4 xx.xxx.xxx.xxx t=0 0 m=audio 25236 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=ptime:20 a=sendrecv [5] 2009/01/08 14:19:02: SIP Rx udp:xx.xx.xxx.xx:41365: ACK sip:2486@xx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK081dc496ab17cc6f From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Contact: <sip:2486@192.168.2.160:5060;transport=udp> Supported: path Authorization: Digest username="2486", realm="our.company.com", algorithm=MD5, uri="sip:911@our.company.com", nonce="9187416fad0b35ed973e718a1e137b40", response="4aefc8322bfd5e97ccd41583d0524294" Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58006 ACK User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 [5] 2009/01/08 14:19:13: SIP Rx udp:xx.xx.xxx.xx:41365: BYE sip:2486@xx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK15d0c4a9ce79d33c From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Supported: path Authorization: Digest username="2486", realm="our.company.com", algorithm=MD5, uri="sip:2486@xx.xxx.xxx.xxx:5060", nonce="9187416fad0b35ed973e718a1e137b40", response="27e6d22da22aff2904d7eb7bf9db8275" Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58007 BYE User-Agent: Grandstream GXP2000 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 [5] 2009/01/08 14:19:13: SIP Tx udp:xx.xx.xxx.xx:41365: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.160:5060;branch=z9hG4bK15d0c4a9ce79d33c;rport=41365;received=xx.xx.xxx .xx From: "2486 2486 (MIKE)" <sip:2486@our.company.com>;tag=0bb7fa864496093b To: <sip:911@our.company.com>;tag=d471b9f82e Call-ID: d6b13bf64de671df@192.168.2.160 CSeq: 58007 BYE Contact: <sip:2486@xx.xxx.xxx.xxx:5060> User-Agent: MIKE/3.1.1.3110 RTP-RxStat: Dur=15,Pkt=752,Oct=129344,Underun=0 RTP-TxStat: Dur=12,Pkt=732,Oct=125904 Content-Length: 0 [5] 2009/01/08 14:19:13: SIP Tx udp:208.71.179.138:5060: BYE sip:911@208.71.179.138 SIP/2.0 Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-99b5921b9df76d3407a2b130263d0ab0;rport From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone>;tag=as5af89d5b Call-ID: dc81ee01@pbx CSeq: 27505 BYE Max-Forwards: 70 Contact: <sip:12486@xx.xxx.xxx.xxx:5060;transport=udp> RTP-RxStat: Dur=15,Pkt=562,Oct=96664,Underun=0 RTP-TxStat: Dur=12,Pkt=583,Oct=100276 P-Asserted-Identity: "2486 2486" <sip:12486@our.company.com;user=phone> Content-Length: 0 [5] 2009/01/08 14:19:13: SIP Rx udp:208.71.179.138:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-99b5921b9df76d3407a2b130263d0ab0;received=xx.xxx.xxx.xxx;rport=5060 From: "2486 2486" <sip:12486@our.company.com;user=phone>;tag=2011644075 To: <sip:911@208.71.179.10;user=phone>;tag=as5af89d5b Call-ID: dc81ee01@pbx CSeq: 27505 BYE User-Agent: 911Enable SBC1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:911@208.71.179.138> Content-Length: 0 [5] 2009/01/08 14:19:13: BYE Response: Terminate dc81ee01@pbx [5] 2009/01/08 14:19:15: SIP Rx udp:xx.xx.xxx.xx:25489: BYE sip:2486@xx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.97:5060;branch=z9hG4bK-20442e27 From: <sip:2486@our.company.com>;tag=2979a3c0d2d91af0o0 To: <sip:15147452143@our.company.com>;tag=7489e2f23f Call-ID: 9f8ab000-e2017bb0@192.168.2.97 CSeq: 103 BYE Max-Forwards: 70 Authorization: Digest username="2486",realm="our.company.com",nonce="ade9fa29951e53261f78d901a54ceb08",uri="sip:2486@xx.xxx.xxx.xxx:5060",algorithm=MD5,response="5e778d6fe7153adcce8c2101215b6264" User-Agent: Linksys/SPA962-6.1.3(a) Content-Length: 0 [5] 2009/01/08 14:19:15: SIP Tx udp:xx.xx.xxx.xx:25489: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.2.97:5060;branch=z9hG4bK-20442e27;rport=25489;received=xx.xx.xxx.xx From: <sip:2486@our.company.com>;tag=2979a3c0d2d91af0o0 To: <sip:15147452143@our.company.com>;tag=7489e2f23f Call-ID: 9f8ab000-e2017bb0@192.168.2.97 CSeq: 103 BYE Contact: <sip:2486@xx.xxx.xxx.xxx:5060> User-Agent: MIKE/3.1.1.3110 RTP-RxStat: Dur=79,Pkt=2655,Oct=662216,Underun=4 RTP-TxStat: Dur=77,Pkt=2642,Oct=665784 Content-Length: 0 [5] 2009/01/08 14:19:15: SIP Tx udp:69.25.128.195:5060: BYE sip:15147452143@69.25.128.204:5064 SIP/2.0 Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK-2d80f78ef41880ef09ef02c6170b78df;rport Route: <sip:69.25.128.195;lr=on> From: "2486 2486" <sip:9879879877@our.company.com;user=phone>;tag=659172529 To: <sip:15147452143@69.25.128.195;user=phone>;tag=as64abbccc Call-ID: e1794021@pbx CSeq: 19490 BYE Max-Forwards: 70 Contact: <sip:9879879877@xx.xxx.xxx.xxx:5060;transport=udp> RTP-RxStat: Dur=79,Pkt=3851,Oct=662372,Underun=5 RTP-TxStat: Dur=77,Pkt=3852,Oct=657708 P-Asserted-Identity: "2486 2486" <sip:9879879877@our.company.com;user=phone> Content-Length: 0 [5] 2009/01/08 14:19:15: SIP Rx udp:69.25.128.195:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;received=xx.xxx.xxx.xxx;branch=z9hG4bK-2d80f78ef41880ef09ef02c6170b78df;rport=5060 From: "2486 2486" <sip:9879879877@our.company.com;user=phone>;tag=659172529 To: <sip:15147452143@69.25.128.195;user=phone>;tag=as64abbccc Call-ID: e1794021@pbx CSeq: 19490 BYE User-Agent: TSG_Global_GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:15147452143@69.25.128.204:5064> Content-Length: 0 [5] 2009/01/08 14:19:15: BYE Response: Terminate e1794021@pbx