Jump to content


  • Posts

  • Joined

  • Last visited

Worm78's Achievements


Newbie (1/14)



  1. 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: (Win32) Broadvox Trunk Thanks, Brian
  2. 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.
  3. 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
  4. 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..."
  5. 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
  6. 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@ 700 00:35 2009/08/19 09:50:15 1000 John Doe (303)
  7. 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@ to [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 [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@ RTP pass-through mode
  8. Worm78

    User Guides

    Yep, Thank You. I searched voicemail flow chart sheet etc. Thanks Again.
  9. 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: (Win32) ;************** ;** Ini File ** ;************** ;Board: MP-118 FXO ;Serial Number: 762960 ;Slot Number: 1 ;Software Version: 5.00A.024 ;Board IP Address: ;Board Subnet Mask: ;Board Default Gateway: ;Ram size: 32M Flash size: 8M ;Num DSPs: 2 Num DSP channels: 8 ;Profile: NONE ;------------------------------ [sYSTEM Params] SyslogServerIP = VXMLFIleName = '' VoiceMenuPassword = 'disable' [bSP Params] PCMLawSelect = 3 LocalOAMIPAddress = 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 = '' 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 = *,,*,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 = 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]
  10. Worm78

    User Guides

    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
  11. 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
  12. 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: REGISTER sip: SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bKb5be3f49 From: <sip:311@>;tag=0019e75aa7c000026a467fec-812307d3 To: <sip:311@> Call-ID: 0019e75a-a7c00002-da86110e-2238b0fd@ 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@;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: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP;branch=z9hG4bKb5be3f49 From: <sip:311@>;tag=0019e75aa7c000026a467fec-812307d3 To: <sip:311@>;tag=82734d6b10 Call-ID: 0019e75a-a7c00002-da86110e-2238b0fd@ CSeq: 101 REGISTER User-Agent: pbxnsip-PBX/ Content-Length: 0 Thanks, Brian
  13. How would you tie a domain to an IP locally?
  14. 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: REGISTER sip: SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK5e6b91a6 From: <sip:311@>;tag=0019e75aa7c00002852f4b58-34de90ae To: <sip:311@> Call-ID: 0019e75a-a7c00002-8c130b80-e2438516@ 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@;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: SIP/2.0 404 Not Found Via: SIP/2.0/UDP;branch=z9hG4bK5e6b91a6 From: <sip:311@>;tag=0019e75aa7c00002852f4b58-34de90ae To: <sip:311@>;tag=5e54e0bea2 Call-ID: 0019e75a-a7c00002-8c130b80-e2438516@ 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></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></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>
  • Create New...