Worm78

I Have 4 new employees staring with us from another company. They each have plain old pots phone lines with call forwarding setup from accross the city. They are each going to call forward their phones to our main sip trunk line. Is there a way for the PBX to detect which line is being forwarded in and then route these calls to each of the 4 extensions based on which line is being forwarded. I think there is just not sure which field this goes in. Version: 3.4.0.3201 (Win32) Broadvox Trunk Thanks, Brian

I'm using the built in tftp on the pbx currently. The SEP_Mac_.cnf / xmls files don't have a registrar or domain field. Only outbound proxy and auth. Its crazy because every other SIP device seems to have the domain field.

Thanks for the replies and the info. My only issue is I have no DNS server on that network. I also have three other companies on the current PBXNSIP setup all in their own domains. Currently just the PBX and a netgear router handing out dhcp on the lan side. Wan goes to a sonicwall firewall. All three use only SNOM phones that have the domain field. I have 8 brand new 7941G phones.May be easier to just sell on ebay and buy Snom. Any ideas on coming in from outside? All four companies are in the same building. Thanks, Brian

Revisiting this issue. I have the latest SIP software upgraded from PBXNSIP. Anything changed in this area? "Well, that means that the PBX could not find that extension. In a single domain environment no problem; just use the name "localhost". However for multiple domains a serious problem, as the Cisco phones seem to have a hard time to use DNS names in the registration. So far the workaround is to use a different IP address for each domain. If someone finds out how to make Cisco phones send DNS names, please let us know..."

Anyone suggest any good SNOM resellers? Good selection and reliable shipping....So on....I have used NewEgg for thelast few but they don't have much of a selection. Thanks

I was told it said that once as well. I have not seen it. I do see it on the log I sent which is really weird. I just tried it and I get 1000. Trunk rolls to a hunt group (700) for 6 seconds which rings the receptionist extension 101, and then goes to the ACD which is 701 I checked the From-Header: on the hunt group and it is set to calling party I didn't see an option on the ACD I will resend the log from 5 minutes ago where I just saw the 1000 come in. EDIT ******** Just founf if I call the other lines coming in it works. Calling the main line it does not. May be a telephone company issue. Grrrrrrrrrrrrrrrr ********* Here is the call log as well 2009/08/19 09:49:13 1000 (1000@192.168.1.253) 700 00:35 2009/08/19 09:50:15 1000 John Doe (303)

Number dialed in form was 985 8299 and .253 is the Audiocodes, 700 is a hunt group. While I'm asking I'm also having an issue where intermittently I get no audio or one way audio (outgoig only) when the system cfwd's to a cell phone. No other issues with the trunk. It seems to happen 1 out of 4 times. I have also had a very garbled sound a few times. [5] 2009/08/17 11:31:53: Identify trunk (IP address and DID match) 1 [7] 2009/08/17 11:31:53: Set packet length to 20 [6] 2009/08/17 11:31:53: Sending RTP for 66993680812102000194319@192.168.1.253#a7b0f0dfd1 to 192.168.1.253:6000 [5] 2009/08/17 11:31:53: Trunk Audiocodes (not global) sends call to account 700 in domain realty [7] 2009/08/17 11:31:53: Looking for EPID 700 [7] 2009/08/17 11:31:53: Set packet length to 20 [6] 2009/08/17 11:31:53: Send codec pcmu/8000 [7] 2009/08/17 11:31:53: Call a99082be@pbx#16196: Clear last request [7] 2009/08/17 11:31:55: Call a99082be@pbx#16196: Clear last INVITE [6] 2009/08/17 11:31:55: Send codec=pcmu/8000 afrer answer [6] 2009/08/17 11:31:55: Sending RTP for a99082be@pbx#16196 to 192.168.1.9:61370 [7] 2009/08/17 11:31:55: Determine pass-through mode after receiving response [7] 2009/08/17 11:31:55: a99082be@pbx#16196: RTP pass-through mode [7] 2009/08/17 11:31:55: 66993680812102000194319@192.168.1.253#a7b0f0dfd1: RTP pass-through mode

TTT

Yep, Thank You. I searched voicemail flow chart sheet etc. Thanks Again.

I have started using an MP118 as a trunk on its own domain. This was used as a backup trunk using the dialplan feature. I added this to its own domain and all works ok for incoming and outgoing calls except CLID. Everything shows up as 1000 or 1001. I have hooked a caller id enabled phone directly to the pots line to ensure the telco is sending info and this works. I see nothing in the logs on the MP118 when calling in that has a phone number. Under end point settings I'm using the automatic dial feature to send it to the trunk ACD which is 700. Caller ID is set to enabled on each fxo. Detect clid from telco is as well enabled End point phone numbers is blank. I'm using three centrex lines and not rolling or using hunt groups on the MP. Device is setup in proxy mode. Any ideas? Thanks, Brian here is the ini file and unit is on latest firmware. PBX Version: 3.4.0.3201 (Win32) ;************** ;** Ini File ** ;************** ;Board: MP-118 FXO ;Serial Number: 762960 ;Slot Number: 1 ;Software Version: 5.00A.024 ;Board IP Address: 192.168.1.253 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 192.168.1.1 ;Ram size: 32M Flash size: 8M ;Num DSPs: 2 Num DSP channels: 8 ;Profile: NONE ;------------------------------ [sYSTEM Params] SyslogServerIP = 10.1.1.89 VXMLFIleName = '' VoiceMenuPassword = 'disable' [bSP Params] PCMLawSelect = 3 LocalOAMIPAddress = 192.168.1.253 RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 [ATM Params] [Analog Params] CallProgressTonesFilename = 'usa_tones_12.dat' [ControlProtocols Params] [MGCP Params] [MEGACO Params] EP_Num_0 = 0 EP_Num_1 = 1 EP_Num_2 = 0 EP_Num_3 = 0 EP_Num_4 = 0 [sS7 Params] [Voice Engine Params] IdlePCMPattern = 85 VoiceVolume = 3 InputGain = 3 DTMFVolume = 0 RFC2833PayloadType = 101 [WEB Params] LogoWidth = '339' [sIP Params] ENABLECALLERID = 1 MAXDIGITS = 11 LOCALSIPPORT = 5060 PLAYRBTONE2IP = 0 REGISTRATIONTIME = 3600 SIPT1RTX = 500 SIPT2RTX = 4000 ISPROXYUSED = 1 SIPDESTINATIONPORT = 5060 PLAYRBTONE2TEL = 2 ISTWOSTAGEDIAL = 0 DETFAXONANSWERTONE = 0 ENABLECURRENTDISCONNECT = 1 CHANNELSELECTMODE = 1 GWDEBUGLEVEL = 5 ENABLERPIHEADER = 1 ENABLEEARLYMEDIA = 1 ISUSERPHONE = 0 SIPSESSIONEXPIRES = 0 SIPGATEWAYNAME = '192.168.1.253' CNONCE = '0a123bcf' PASSWORD = '787899' PRACKMODE = 1 SIPMAXRTX = 7 ASSERTEDIDMODE = 0 ISUSERPHONEINFROM = 0 ADDTON2RPI = 1 USESOURCENUMBERASDISPLAYNAME = 1 MINSE = 90 IPALERTTIMEOUT = 180 ISFAXUSED = 1 SIPTRANSPORTTYPE = 0 TCPLOCALSIPPORT = 5060 RINGSBEFORECALLERID = 2 TLSLOCALSIPPORT = 5061 ENABLESIPS = 0 USERAGENTDISPLAYINFO = '' SESSIONEXPIRESMETHOD = 0 USEDISPLAYNAMEASSOURCENUMBER = 0 USETELURIFORASSERTEDID = 0 USESIPTGRP = 0 SIPSUBJECT = '' CODERNAME = g711Ulaw64k,20,0,$$,0 PREFIX = *,192.168.1.254,*,0,255 TARGETOFCHANNEL0 = 700,1 TARGETOFCHANNEL1 = 700,1 TARGETOFCHANNEL2 = 700,1 TARGETOFCHANNEL3 = 700,1 TARGETOFCHANNEL4 = 700,1 TARGETOFCHANNEL5 = 700,1 TARGETOFCHANNEL6 = 700,1 TARGETOFCHANNEL7 = 700,1 TRUNKGROUP = 1-1,,0 TRUNKGROUP = 2-2,,0 TRUNKGROUP = 3-3,,0 TRUNKGROUP = 4-4,,0 TRUNKGROUP = 5-5,,0 PROXYIP = 192.168.1.254 TXDTMFOPTION = 4 ENABLECALLERID_0 = 1 ENABLECALLERID_1 = 1 ENABLECALLERID_2 = 1 ENABLECALLERID_3 = 1 ENABLECALLERID_4 = 1 ENABLECALLERID_5 = 1 ENABLECALLERID_6 = 1 ENABLECALLERID_7 = 1 [VXML Params] [iPsec Params] [Audio Staging Params] [PSTN-SDH Params]

Not sure if this is the correct area for this but I'm looking for user guides for the voicemail. I have searched the wiki and the new support site with no luck. I'm looking for the simple guide that hangs on cubicle walls. Anyone have one? Can you link me or send an email to elwormo @ hotmail .com

Jlumby, Thanks for the help. I heard you were the god of cisco in multi domain. Are you starting with an xml file by adding a the mac under the extension ( bind to mac) then making the changes below on the generated file. Then naming it with the correct name and adding to the tftp directory? Can you explain on this one? "SRV records created for the domain" I'm concenred I may have a DNS issue as if I ping the host name of the PBX from the server I get the external IP of the server. My lan connection has no DNS settings listed. I have a small netgear router sedning out itself as DNS since it is doing DHCP. I will add iteself and hand out dns. Hopefully that won't mess up my snoms in my other working domains. Thanks for the help, Brian

After trying that I don't even see sip registration in the log. Must not like it. If I keep cisco phones on only 1 of my 4 domains could I name that one doamin local host or the IP of the sever? Maybe just add an alias. I could then use snom on the other domains. My other three domains are done and working its just this one division I'm adding a domain with cisco only phones. Or would this make my other snoms "Flip Out"? I quickly tried this and here iss the log. 7] 2009/07/24 10:16:09: SIP Rx udp:192.168.1.27:49155: REGISTER sip:192.168.1.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bKb5be3f49 From: <sip:311@192.168.1.254>;tag=0019e75aa7c000026a467fec-812307d3 To: <sip:311@192.168.1.254> Call-ID: 0019e75a-a7c00002-da86110e-2238b0fd@192.168.1.27 Max-Forwards: 70 Date: Tue, 05 May 2009 20:34:24 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7941G/8.5.2 Contact: <sip:311@192.168.1.27:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0019e75aa7c0>";+u.sip!model.ccm.cisco.com="115" Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP0019E75AA7C0 Load=SIP41.8-5-2S Last=initialized" Expires: 3600 [7] 2009/07/24 10:16:09: SIP Tx udp:192.168.1.27:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bKb5be3f49 From: <sip:311@192.168.1.254>;tag=0019e75aa7c000026a467fec-812307d3 To: <sip:311@192.168.1.254>;tag=82734d6b10 Call-ID: 0019e75a-a7c00002-da86110e-2238b0fd@192.168.1.27 CSeq: 101 REGISTER User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Length: 0 Thanks, Brian

How would you tie a domain to an IP locally?

Multi domain and the file is being created under <working dir>/generated/<domain>/<extension> The file created is named sep_cnf.xml no mac in the middle. I tired to change the file name to SEPmacaddress.cnf.xml I rebooted the phone and it will create a new file named sep_cnf.xml in the same folder. Name and password look correct in file. May be having the same issue as this guy. No confirmed resolution posted. I have tried his suggestion on the generated fileb ut it is changed back. I also copied this file, edited his suggested changes, named it correctly and added it to the tftp folder. No luck If I do try the manual method does anyone know where domain or registrar info is in regards to the xml file, my snom phones call it a registrar intheri xml but this is not an option in cisco files. http://forum.pbxnsip.com/index.php?showtopic=367 Here is the phone trying to register. [7] 2009/07/23 09:21:15: SIP Rx udp:192.168.1.27:49155: REGISTER sip:192.168.1.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK5e6b91a6 From: <sip:311@192.168.1.254>;tag=0019e75aa7c00002852f4b58-34de90ae To: <sip:311@192.168.1.254> Call-ID: 0019e75a-a7c00002-8c130b80-e2438516@192.168.1.27 Max-Forwards: 70 Date: Tue, 05 May 2009 20:34:26 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7941G/8.5.2 Contact: <sip:311@192.168.1.27:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0019e75aa7c0>";+u.sip!model.ccm.cisco.com="115" Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP0019E75AA7C0 Load=SIP41.8-5-2S Last=initialized" Expires: 3600 [7] 2009/07/23 09:21:15: SIP Tx udp:192.168.1.27:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK5e6b91a6 From: <sip:311@192.168.1.254>;tag=0019e75aa7c00002852f4b58-34de90ae To: <sip:311@192.168.1.254>;tag=5e54e0bea2 Call-ID: 0019e75a-a7c00002-8c130b80-e2438516@192.168.1.27 CSeq: 101 REGISTER Content-Length: 0 [7] 2009/07/23 09:21:17: Open TFTP port 2018 [6] 2009/07/23 09:21:17: TFTP: Request dialplan.xml XML file pasted below: <device xsi:type="axl:XIPPhone" ctiid="1566023366"> <deviceProtocol>SIP</deviceProtocol> <sshUserId>admin</sshUserId> <sshPassword>admin</sshPassword> <devicePool> <dateTimeSetting> <dateTemplate>D-M-YA</dateTemplate> <timeZone>Eastern Standard/Daylight Time</timeZone> <ntps> <ntp> <name>pool.ntp.org</name> <ntpMode>Unicast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort></securedSipPort> </ports> <processNodeName>192.168.1.254</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <sipProfile> <sipProxies> <backupProxy></backupProxy> <backupProxyPort>5060</backupProxyPort> <emergencyProxy></emergencyProxy> <emergencyProxyPort>5060</emergencyProxyPort> <outboundProxy></outboundProxy> <outboundProxyPort>5060</outboundProxyPort> <registerWithProxy>true</registerWithProxy> </sipProxies> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI> <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI> <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI> <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI> <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI> <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI> <rfc2543Hold>false</rfc2543Hold> <callHoldRingback>2</callHoldRingback> <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock> <callerIdBlocking>2</callerIdBlocking> <dndControl>0</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <sipInviteRetx>6</sipInviteRetx> <sipRetx>10</sipRetx> <timerInviteExpires>180</timerInviteExpires> <timerRegisterExpires>3600</timerRegisterExpires> <timerRegisterDelta>5</timerRegisterDelta> <timerKeepAliveExpires>120</timerKeepAliveExpires> <timerSubscribeExpires>120</timerSubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects> <remotePartyID>false</remotePartyID> <userInfo>None</userInfo> </sipStack> <autoAnswerTimer>0</autoAnswerTimer> <autoAnswerAltBehavior>false</autoAnswerAltBehavior> <autoAnswerOverride>true</autoAnswerOverride> <transferOnhookEnabled>false</transferOnhookEnabled> <enableVad>false</enableVad> <preferredCodec>g711</preferredCodec> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <natEnabled>false</natEnabled> <natAddress></natAddress> <phoneLabel>311</phoneLabel> <stutterMsgWaiting>1</stutterMsgWaiting> <callStats>false</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>311</featureLabel> <proxy>192.168.1.254</proxy> <port>5060</port> <name>311</name> <displayName>Cordless Phone</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>311</authName> <authPassword>123456789123</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>1</messageWaitingLampPolicy> <messagesNumber>311</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>311</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> </sipLines> <voipControlPort>5060</voipControlPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> </sipProfile> <commonProfile> <phonePassword>pbxnsip</phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>2</callLogBlfEnabled> </commonProfile> <loadInformation>SIP41.8-5-2S</loadInformation> <vendorConfig> <disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <pcPort>1</pcPort> <settingsAccess>1</settingsAccess> <garp>0</garp> <voiceVlanAccess>0</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <webAccess>1</webAccess> <spanToPCPort>1</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer></loadServer> </vendorConfig> <versionStamp></versionStamp> <networkLocale></networkLocale> <networkLocaleInfo> <name>United_States</name> <version>5.0(2)</version> </networkLocaleInfo> <deviceSecurityMode>1</deviceSecurityMode> <authenticationURL></authenticationURL> <directoryURL></directoryURL> <idleURL></idleURL> <informationURL></informationURL> <messagesURL></messagesURL> <proxyServerURL></proxyServerURL> <servicesURL></servicesURL> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>4</transportLayerProtocol> <capfAuthMode>0</capfAuthMode> <capfList> <capf> <phonePort>3804</phonePort> </capf> </capfList> <certHash></certHash> <encrConfig>false</encrConfig> </device>

I grabbed a new phone from the box and it seems to be pulling tftp firmware updates. Waiting to see if the phone config files come over. Phone booted SIP, but not registering. Phone says registering over and over. Here is log. Wireshark shows the phone tring to access the server IP for a bit with a SIP 404 error. In web gui there is nothing under registration popping up and the log doesn't show any registering other then what is below. [6] 2009/07/22 17:41:58: TFTP: File CTLSEP0019E75AA7C0.tlv not found [6] 2009/07/22 17:41:59: TFTP: Request SEP0019E75AA7C0.cnf.xml [7] 2009/07/22 17:41:59: UDP: Opening socket on 0.0.0.0 [7] 2009/07/22 17:41:59: Open TFTP port 1589 [8] 2009/07/22 17:41:59: TFTP: Transfer finished successfully [6] 2009/07/22 17:42:08: TFTP: File /mk-sip.jar not found [7] 2009/07/22 17:42:13: UDP: Opening socket on 0.0.0.0 [7] 2009/07/22 17:42:13: Open TFTP port 1590 [6] 2009/07/22 17:42:13: TFTP: Request dialplan.xml [8] 2009/07/22 17:42:13: TFTP: Transfer finished successfully

Any comment on the 0.0.0.0 address? Is that normal and can it work with this being listed? I have had the mac address under a user and it creates a folder in generated labeled with the correct ext and domain. Fails to pull files from the tftp location. FYI System is on its own network with dual nics. A small rounter has its lan side plugged in to handle dhcp only. Lan (internal) side of pbx has no gateway. If anything is entered in the field my trunks go down. Wan is currently plugged directly to public switch.

Mac starts with 0019E7 I have the log set to 8 and have tftp only checked and get nothing. This has always been the case. I don't think tftp is working. What did you think of the log file showing 0.0.0.0 for the tftp IP?

I'm working with a csico 7941 on a multi domain windows 3.4 server Currently the system is all snom phones and i'm trying to add a few cisco phones. I have TFTP working on another test PC (not a pbx system) and I have upgraded the firmware to SIP and tested some xml config files. All seems to be working. Problem is once moved to the PBX tftp isn't working. I believe all the correct files are in place. Wondering if this is caused by no gateway on the lan nic? If so can I add the pbx internal IP as its gateway with no other issues? On the positive side if I add the mac address to an extension the system is generating a cfg file in the proper domain folder. next question is once I have TFTP working should I fear all my old Snom phones that were manually setup failing? I just upgraded to 3.4 and have not used PNP at all in the past. Thanks, Brian Here is the log after reboot. [1] 2009/07/21 21:11:49: Starting up version 3.4.0.3201 [7] 2009/07/21 21:11:49: Found time zones HST AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET GMT+2 GMT+3 GMT+4 GMT+5 GMT+6 GMT+7 GMT+8 GMT+9 CST CAT IST AUS1 AUS2 AUS3 AUS4 AUS5 AUS6 GMT [1] 2009/07/21 21:11:49: Working Directory is C:\Program Files\pbxnsip\PBX [7] 2009/07/21 21:11:49: UDP: Opening socket on 0.0.0.0 [5] 2009/07/21 21:11:49: Starting threads [7] 2009/07/21 21:11:49: UDP: Opening socket on 0.0.0.0 [7] 2009/07/21 21:11:49: TCP: Opening socket on 0.0.0.0:80 [7] 2009/07/21 21:11:49: TCP: Opening socket on 0.0.0.0:443 [7] 2009/07/21 21:11:49: UDP: Opening socket on 0.0.0.0:161 [7] 2009/07/21 21:11:49: UDP: Opening socket on 0.0.0.0:69 [7] 2009/07/21 21:11:49: UDP: Opening socket on 0.0.0.0:5060 [7] 2009/07/21 21:11:49: TCP: Opening socket on 0.0.0.0:5060 [7] 2009/07/21 21:11:49: TCP: Opening socket on 0.0.0.0:5061

I'm trying to authorize 6 cisco phones on a multi domain pbx. I'm using tftp to push the config file to the phone with the mac specified file. I have dial plan and ringtone file in as well. Phones have been updated to sip 8 3 1. I used the wiki config file as a guide, but i'm thinking I have something wrong in the proxy / registar area. Does anyone have a known good conf file for our PBX? I'm running 3.1.1.3110 (Win32) Also is there a way to use a web gui to make changes in the latest firmware from cisco? I'm still waiting on my smartnet from cdw to go through to get the latest software. I normally run snom 300 phones and setup proxy server to pbx IP, registar to domain name, and account as extension and all is well. Just not sure what xml tag is for the domain, if multi domain is supported. I have 4 domains currently.

I have it setup as listed above in the a b c d scenario and all seems to be working. No trunks are global. Each trunk has its own DID. In use by the users now. Hoping for no issues.

I did not plan to have a global trunk. Maybe I need to. My thought was have two domains with their own trunks. In the trunk settings of each domain have the same info for Broadvox in both, Only have something sending the correct DID to the correct trunk / Attendant Extension. If I can do this what field would I put the phone number in on trunk settings. Currently they are both setup and work outgoing only. I have not renamed my localhost at this point if that matters. All incoming calls go to local host trunk. Should I use one trunk in domain A and then set it to global and send DID to proper domain attendants globally. If so where would the DID from Domain A's trunk go in the auto attendant of each domain? Diagram below... Current Domain A (currently named localhost), has audio codes(DID Z) and broadvox trunk using DID X or Y. This domain only needs audiocodes GW DID Z Domain B has broadvox trunk, outbound only currently, needs DID X to roll to attendant in domain B Domain C has broadvox trunk, outbound only currently, needs DID Y to roll to attendant in domain C Domain D has Teliax trunk that is working. I have 2 numbers from Broadvox

I currently have the localhost domain setup with 30 users. 15 are leaving I have 5 new coming in to each of the 3 new domains. Those are setup and working as well internally. I think from what you are saying I will need to rename the localhost domain, let call it A. Then go to each of the 15 phones and add the domain name in the reg field. No biggy since this is on a small scale and all local. My main concern is then the trunk settings. I will set all to non-global. Trunk A will use its current audiocode with copper coming in for that domain. It is working fine and set to the localhost right now. I have domain B with a trunk from Teliax. I'm working on setting it up now as it is moving from another public IP. i just need to call Teliax and update this info to register. This should be easy enough. My concern is I have domain C and D both with numbers on the same account from broadvox. Is there a field in the trunk settings I use to let broadvox or the pbx know which number to bring into that domain? I have reached out to Broadvox to see if they send the number in the sip header or if this is something we need to request. I'm guessing they do. 123-456-1111 to one domain and 123-456-2222 to other domain?

Just curious, Can you have multiple domains on one server with the same extensions on each domain on the same physical network? I would assume each will needs it own trunk in its own domain. My main stump would be when registering my sip phones how will it know the difference between user 301 on one domain and user 301 on the other. I have multiple DID numbers currently with one domain and one trunk. All users in one big phone system. I'm assuming I can take DID A and send to ACD for domain A. Then setup trunk on domain B and send it to ACD B. Essentially running 4 small companies on one 50 user PBX. Registering: I'm thinking use Domain name where Registar is and outbound proxy as IP of server. My main concern is how to take a DID to the correct trunk. or how will provider (Currently using Broadvox) look at multiple connections coming in. Version: 3.1.1.3110 (Win32) Snom 300 and 320's Quick update, I was tinkering and added a test domain, setup an extension unused by the other domain and attempted to setup the broadvox trunk. Added IP of box in the exact way as the other trunk currently in use on my original domain. It has been a while since I set it up but I think there is no login and pass it just uses our IP/MAC . I just need to figure out where to put the DID number to get it to go the correct domain/ext. I did get the other extension with the same extension number used in another domain working with the method listed above using domain name in the registar field. Works great.

Issue Resolved: Cable modem had ARP Storm detection enabled. This caused a 5-10 second period of one way audio. Changed cable modems and issue went away. Reinstalled and provider turned setting off and issue went away. This was an old Zenith cable modem. Thanks, Brian