WHAT IS A SOFTPHONE ??

As the word Softphone only appears under the "Phone" headings, and then, only how to set it up, without any explanation as to what it is, and how it works, as compared with a Digital (normal) phone or Hub Phone.
The Hub Phone does appear around "Broadband" headings and is explained what it is and how it works and fearures.
However, when i type "What is a Softphone" into any phone search boxes i get a Nil Find !!!
So please, WHAT IS A SOFTPHONE, how is it different from other phones, why should one use a SOFTPHONE ?
PS. An amendment to the search facility is required when "What is a Softphone" is typed into the box.......an explanation should somehow appear.
Or am I the only one that can't find out on the BT site what a Softphone is ???
Solved!
Go to Solution.

Softphone is a software application which runs on your computer and allows you to send and receive phone calls, just like Skype.
BTs version allows you to use you Broadband Talk number for this purpose. You download the Softphone application, and, provided you have a microphone and speaker on your computer, you can use it as a telephone.
It is useful if you are away from home and have a laptop with you. You can use it to make phone calls.
There are some useful help pages here, for BT Broadband customers only, on my personal website.
BT Broadband customers - help with broadband, WiFi, networking, e-mail and phones.

Similar Messages

Best Headsets for SoftPhone

What are the best sound quality headsets (monaural wired, USB, monaural wireless, USB, monaural wireless, Bluetooth)? I've tried some that sound ok, and some that sound just horrible. And some Bluetooth Plantronics devices are not even seen by the SoftPhone.
BTW - we are still running CM 3.2(2c) (for now).

Hi Mike,
Headsets are of course a very personal "choice" type of item. We use Plantronics headsets and our users really like them. What version of softphone are you using? Here is a link to the supported Plantronics headsets for Cisco Softphone 1.3
http://www.plantronics.com/canada/en_US/productfinder/productlist.jhtml?refresh=all&mftr=CISCO%20SYSTEMS&_requestid=579398
There is also this new DA40 adaptor that will allow you to use a whole range of Plantronics headsets.
" The DA40 connects any Plantronics professional headset (for example, SupraPlus® headsets) to any softphone using the computers USB port providing customers with an attractive alternative to consumer-grade headsets."
From this doc;
http://www.plantronics.com/canada/en_US/press/releases/index.jhtml?id=pr_20051021_1828_en_US
We personally like the H-171 headset.
Hope this helps!
Rob
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What type of headset with microphone can be used for softphone on ipad?

what type of headset with microphone can be used for softphone on ipad? ideally wireless, if not a simple one not too bulky

I use the same bluetooth headset that I use with my iphone, Jabra.
and in using skype or other voip apps, it works just like my iphone does.

What are the setting in order to use headset with vontage softphone, caller can't hear me

have dv7 entertainment notebook with IDT audio. When I use vontage softphone caller can not hear me but I can hear them. Is there some default setting for the notebook that I am missing or is it a vontage issue?
windows 7
HP Pavilion dv7 Notebook PC
Intel i5
64 bit os

I had the same issue. It was not being muted by my face (although sometimes that would happen). For me you could actually tap on the back of the phone and get it to work...but only for a short period of time.
I went through all the normal trouble shooting items... rebooting... restoring...etc. At the end of the day I took it into Apple store. They said they've had other people with the same issue and the best bet is to replace the phone while it's still under warranty. You have one year from purchase date unless you bought the extended Applecare.
I wouldn't bother with too much more trouble shooting...not since this is a known issue that others have had happen.

Jabber Client softphone is "No Connected"

Hi,
I am trying to setup a lab for cisco partner testing. I created Collab 10.5 topology with cucm & cups 10.5. I am running out of ideas of what could be wrong. My jabber client for windows works with deskphone but it does not work with it's softphone (CSFXXXXX). Yes, i have CCMCIP profile created, user & device associated but still does not work both within lab network or external network
I am seeing the error below when I do a show connection status for the softphone - see attachment
Connection error. Ensure the server information in the Phone Services tab on the Options window is correct.
I do not have the "Phone Accounts" under options. I even uploaded jabber-config.xml file to tftp server and still no phone accounts tab. I can ping the cti hostname & ip from the pc i am working.
SOS
TIA
Jeya

Actually, I read it on a thread in this community. I just went thru’ the deployment guide and I saw that section too. I have uploaded it to the tftp server and no change for the phone tabs. Only the Directory service becomes online after uploading that file but no change on the phone account tabs. Below is the contents of the jabber-config.xml file that I generated using the file generator tool:
<?xml version="1.0" encoding="utf-8"?>
<config version="1.0">
<Options>
   <StartCallWithVideo>false</StartCallWithVideo>
   <Start_Client_On_Start_OS>true</Start_Client_On_Start_OS>
</Options>
<Phone>
   <TftpServer1>hq-cucm-pub.abc.inc</TftpServer1>
   <CtiServer1>hq-cucm-pub.abc.inc</CtiServer1>
   <CcmcipServer1>hq-cucm-pub.abc.inc</CcmcipServer1>
</Phone>
<Policies>
   <EnableSIPURIDialling>true</EnableSIPURIDialling>
   <DirectoryURI>mail</DirectoryURI>
   <ServicesDomainSsoEmailPrompt>OFF</ServicesDomainSsoEmailPrompt>
</Policies>
<Presence>
   <PresenceServerAddress>hq-cups.abc.inc</PresenceServerAddress>
</Presence>
<Directory>
   <ConnectionType>1</ConnectionType>
   <PrimaryServerName>directory.abc.inc</PrimaryServerName>
   <BDIPrimaryServerName>directory.abc.inc</BDIPrimaryServerName>
   <ServerPort1>389</ServerPort1>
   <BDIServerPort1>389</BDIServerPort1>
   <UseWindowsCredentials>0</UseWindowsCredentials>
   <ConnectionUsername>Administrator</ConnectionUsername>
   <BDIConnectionUsername>Administrator</BDIConnectionUsername>
   <ConnectionPassword>1vtG@lw@y</ConnectionPassword>
   <BDIConnectionPassword>1vtG@lw@y</BDIConnectionPassword>
</Directory>
</config>
Thks!
Jeya

Softphone Registration Error - Jabber 9.2.5 with CUCM/CUPS 8.6

All,
I'm having trouble getting my softphone feature inside Jabber 9.2.5 to work with my CUCM/CUPS 8.6 deployment. I have the CCMIP profile setup and user associations in place. The show connection status indicates a status of Not Connected with an error of "Connection error. Ensure the server information in the Phone Services tab on the Options window is correct. Contact you system administrator for assistance. The CCMCIP address is correct.
Any ideas?
I verified the service is running in CUCM. The Jabber client appears to never have attempted to register to CUCM. Status of the endpoint in CUCM is unknown, unknown.
Thanks!
Jason

Sorry for the delay in getting back to you. You can determine what CUCM server "name" the server is trying to use by doing a TFTP GET of the device config file from a workstation with access to the TFTP server.
An example from a windows box is
tftp -i GET .cnf.xml
This will pull a copy of the device's config file to the pc. You can then open it with any text editor and look for the CUCM name. It is most likely the hostname of the subscriber in question. If you can't resolve the name by itself (no fqdn) then the reg will fail. At least it did in my case.
You can change the server name to IP by browsing to System > Server in CUCM. Change the device in question name to the correct IP.
Alternatively you could correct the name resolution issue.
HTH,
Jason

Jabber for Windows 9.2.4 deskphone/softphone no connection

I have an issue with what appears to be only my desktop. If I am on VPN (anyconnect client) Jabber phone control and softphone works perfectly. If I am hardwired to my ASA and coming across my L2L IPSEC tunnel IM and Presence withing Jabber works but I have a red X through the phone on my Jabber client and cannot switch to Softphone mode (option doesn't do anything when I click on it). I have other users running the same setup and they work fine, other internal users can run Jabber without issue as well. I have full connectivity to the IM&P and CUCM TFTP, CCMCIP, and CTI servers. any ideas?

If you have other users on the same VPN setup, running on the same wired connection the question that comes to mind is if any PC-side firewall/HIDS exceptions didn't get opened for Jabber?
Please remember to rate helpful responses and identify helpful or correct answers.

SoftPhone over VPN audio issue

Having a "one-way" audio issue when IPC connects over VPN to any Main Office IP Phones (7960). Remote IPC (softphone) shows "0 Received" Packets, but the IP Phone shows them as being sent.
This does NOT happen if IPC contacts another IPC on a PC at Main Site, or any kind of phone at the other branch offices.
Have looked at all of the audio settings on the VPN IPC unit, they are correct.
Any ideas?

Try to sniff the Ethernet traffic during call setup at the CallManager. When Callmanager sends Call Control messages to remote party, it could send wrong IP address or UDP port as an RTP parameters.
Another components could block the traffic are firewalls. How do you have your VPN tunnel built? Do you use Cisco VPN Client? What terminates your VPN traffic at the central location? PIX, VPN3000 or Router? Traffic may go one way without a change, and some protocol FixUp may be triggered for this traffic on the way back. If you have access to firewall logs, check if you have any traffic dropped.
Good luck,
Mike
http://www.headsetadapter.com

Cisco Jabber Softphone not working

Hi
The softphone is not working for cisco jabber windows
Attached is the logs
CUCM AND CUPS version 9.x
I found the below but is not clear to me what is the issue
2013-10-10 12:53:42,577 DEBUG [0x00001360] [tionplugin\StatusBarPhoneButton.cpp(127)] [plugin-runtime] [OnCanSelectDeskphone] - enabling deskphone menu items
2013-10-10 12:53:42,577 DEBUG [0x000011a8] [src\config\TftpHelper.cpp(114)] [csf.ecc] [ecc::TftpHelper::doRetrieveFile] - Received error: eFileNotFound downloading file using HTTP from 'http://10.255.0.11:6970/CTLSEPCSFKIKACHRISTOU.tlv', not trying TFTP following an explicit HTTP "not found" error.

Hi Chrysostomos,
I assumes your device name is CSFKIKACHRISTOU
You have added an ower id t the Device
You have allowed CTI Control for the device
You have associated the device to your user in End User Page under Device association.
Your user has is a member of both Standard CCM End Users and Standard CTI Enabled groups.
Now if you are using as softphone, the jabber client retrieves the TFTP server IP Addresses from IMP. Once it has the address, the client download the CSF config file direct from the TFTP Server.
Have you configured the TFTP Servers in IMP?
Application->Legacy Clients->Settings --- specify primary and backup tftp servers.
Exit from Jabber completely and re-login.
See how that goes.
Ben
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Cisco Jabber for Windows unable to make calls in softphone mode

Hi,
I am facing an issue when I make a call from jabber in softphone mode to another jabber client my call fails and using wireshark traces I see SIP2.0 503 Service Unavailable.
I have followed the guide and was able to deploy it once before successfully. This is a lab environment and we don't have any CSS or partitions.
If someone can please guide me :)
Regards
Hassan

Hi Hassan,
I found four calls in the trace file with the same calling/called numbers. I analyzed one of them.
Calling party - 42001
Called party - 42002
00267676.002 |17:15:14.335 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 122.129.75.131 on port 23600 index 689 with 2610 bytes:
[7525,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.60:54441;branch=z9hG4bK00005d94
From: "42001" <sip:[email protected]>;tag=3c77e6734e8e00180000789e-000014f2
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Tue, 21 Apr 2015 11:29:56 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CSF
Contact: <sip:[email protected]:54441;transport=tcp>;video;bfcp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "42001" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 1469
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 5552 0 IN IP4 192.168.1.60
s=SIP Call
b=AS:4000
t=0 0
m=audio 21996 RTP/AVP 9 104 105 0 8 18 101
c=IN IP4 192.168.1.60
a=rtpmap:9 G722/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
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 0-15
a=sendrecv
m=video 25898 RTP/AVP 126 97
c=IN IP4 192.168.1.60
b=TIAS:4000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42E01F;packetization-mode=1;level-asymmetry-allowed=1;max-fs=3601;max-rcmd-nalu-size=32000
a=imageattr:126 recv [x=[32:1:1280],y=[18:1:720],par=1.7778,q=1.00]
a=content:main
a=label:11
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42E01F;packetization-mode=0;level-asymmetry-allowed=1;max-fs=3601
a=imageattr:97 recv [x=[32:1:1280],y=[18:1:720],par=1.7778,q=1.00]
a=sendrecv
m=video 32652 RTP/AVP 126 97
c=IN IP4 192.168.1.60
b=TIAS:4000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42E01F;packetization-mode=1;level-asymmetry-allowed=1;max-fs=4099;max-rcmd-nalu-size=32000
a=content:slides
a=label:12
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42E01F;packetization-mode=0;level-asymmetry-allowed=1;max-fs=4099
a=sendrecv
m=application 5070 UDP/BFCP *
c=IN IP4 192.168.1.60
a=floorctrl:c-s
a=confid:4
a=floorid:2 mstrm:12
a=userid:4
a=setup:actpass
a=connection:new
a=sendrecv
00267702.007 |17:15:14.342 |AppInfo |Digit analysis: match(pi="2", fqcn="42001", cn="42001",plv="5", pss="", TodFilteredPss="", dd="42002",dac="0")
00267702.008 |17:15:14.342 |AppInfo |Digit analysis: analysis results
00267702.009 |17:15:14.342 |AppInfo ||PretransformCallingPartyNumber=42001
|CallingPartyNumber=42001
|DialingPartition=
|DialingPattern=42002
00267709.001 |17:15:14.343 |AppInfo |LBMIF: CI: 30582939 ASSOC    30582940
00267709.002 |17:15:14.343 |AppInfo |LBMIF: CI: 30582940 ASSOC'   30582939
00267715.000 |17:15:14.343 |SdlSig   |CcSetupReq                             |restart0                       |LineControl(1,100,174,27)        |Cdcc(1,100,219,42)               |1,100,14,10382.36^122.129.75.131^CSFHASSANRAZI |[R:N-H:0,N:2,L:0,V:0,Z:0,D:0] CI=30582940 CI.branch=0 sBPL.plid=65 sBPL.l=1 sBPL.pl=5 sBPL.msd=0 FDataType=0opId=0ssType=0 SsKey=0invokeId=0resultExp=Fbpda=F pi.piid=30 pi.l=0 pi2.piid=30 pi2.l=0 pi3.piid=30 pi3.l=0 FQCGPN=ti=1nd=42001pi=0si1 preXCgpn=tn=0npi=0ti=1nd=42001pi=0si1 cgPart= cgPat=42001 cgpn=tn=0npi=0ti=1nd=42001pi=1si1 cgpnVM= unXCgpn=tn=0npi=0ti=1nd=42001pi=1si1 cName=locale: 1 Name: UnicodeName: pi: 1 DD=tn=0npi=1ti=1nd=42002User=42002Host=192.168.201.101Port=5060PassWord=Madder=Transport=4mDisplayName=RawUrl=sip:[email protected];user=phoneOrigPort=0pi=0si1 origDD=tn=0npi=1ti=1nd=42002User=42002Host=192.168.201.101Port=5060PassWord=Madder=Transport=4mDisplayName=RawUrl=sip:[email protected];user=phoneOrigPort=0pi=0si1 preXCdpn=tn=0npi=0ti=1nd=42002pi=0si0 preXTagsList=SUBSCRIBER preXPosMatchList=42002 cdPart= cdPat=42002 cdpn=tn=0npi=0ti=1nd=42002pi=1si1 cdpnVMbox= localPatternUsage=2 connectedPatternUsage=2 itrPart= itrPat= LRPart= LRPat=42002 LR=tn=0npi=0ti=1nd=42002pi=0si1 LRVM= LRName=locale: 1 Name: UnicodeName: pi: 0 FQOCpdn=ti=1nd=42002pi=0si1 fFQLRNum=ti=1nd=42002pi=0si1 oPart= oPat=42002 oCpdn=tn=0npi=0ti=1nd=42002pi=0si1 oCdpnVM= oRFR=0 oName=locale: 1 Name: UnicodeName: pi: 0 ts=SUBSCRIBER posMatches=42002 withTags= withValues= rdn.l=0IpAddrMode=0 ipAddrType=0 ipv4=122.129.75.131:23600 region=Default capCount=7 ctiActive=F ctiFarEndDev=1 ctiCCMId=1 cgPtyDev=CSFHASSANRAZI callInst=0 confCallInst=0 OLF=1Supp DTMF=3DTMF Cfg=1DTMF Payload=101isOffNetDev=F bc.l=4 bc.itr=8 bc.itc=8 bc.trm=6 bc.tm=16 maxForwards=69 cgpnMaskedByRedirect=F callingDP=1b1b9eb6-7803-11d3-bdf0-00108302ead1 featCallType=0 callingUserId= UnicodeName: muteEnabled=0 associatedCallCI=0 featurePriority=1 nonTargetPolicy=0 unconsumedDigits= suppressMOH=F numPlanPkid =6807b55c-7ab5-4f8d-8fba-db383ecfaaa4 networkDomain= bitMask=0 SetupReason=0 routeClass=1 sideACmDeviceType=4 protected=1 ControlProcessType=0 tokens=0 isPresent=F transitCount=0 geolocInfo={geolocPkid=, filterPkid=, geolocVal=, devType=4} locPkid=29c5c1c4-8871-4d1e-8394-0b9181e8c54d locName=Hub_None deductBW=F fateShareId=StandAloneCluster:30582939 videoTrafficClass=Desktop oFromAnalogDvc=F bridgeParticipantID= callingUsr= remoteClusterID= isEMCCDevice=F lHPMemCEPN= cHPMemCEPN= uri=ti=1User=Host=Port=0PassWord=Madder=Transport=4mDisplayName=RawUrl=<sip:[email protected]:54441;transport=tcp>OrigPort=0pi=0si1 param=;video;bfcp M=Unknown ;rc=26624 Hdrs= CanSupportSIPTandN=true TransId=0 AllowBitMask=0x7bf UserAgentOrServer=Cisco-CSF OrigDDName=locale: 1 Name: UnicodeName: pi: 0 mCallerId= mCallerName=LatentCaps=null icidVal= icidGenAddr= oioi= tioi= ptParams= receivedPAID= routeHdr= routeCepn= requestURI= PCVFlag=F originallyHadISUP=F isIMSFinalRoute=F IMSMode=0 SideABibEnabled= 0 isCgpnNonPreemptable=F isCdpnNonPreemptable=F origDP=1b1b9eb6-7803-11d3-bdf0-00108302ead1 lastRedirectingDP=1b1b9eb6-7803-11d3-bdf0-00108302ead1 originalLRG= lastRedirectingLRG= nwLoc=0 rstr= FarEndDeviceName=CSFHASSANRAZI hdrMOH=0 CAL={v=ffffffff, m=ffffffff, tDev=F, res=F, devType=0}
00267715.001 |17:15:14.343 |AppInfo |LineControl(27) - 0 calls, 0 CiReq, busyTrigger=2, maxCall=6
00267715.002 |17:15:14.343 |Created |                                       |                               |LineCdpc(1,100,175,74)           |LineControl(1,100,174,27)        |                                         |NumOfCurrentInstances: 2
00267758.001 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 55 (SIPSPI_EV_SEND_FAILURE_MSG)
00267758.002 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0xe5136ad8/ccsip_spi_process_event: Send Error for event(0xe5139420)
00267758.003 |17:15:19.364 |AppInfo |//SIP/Stack/Error/0x0/act_idle_send_msg_failure: Send Error to 192.168.1.172:54901 for transport TCP
00267758.004 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0xe5136ad8/ccsip_set_cc_cause_for_spi_err: Categorized cause:38, category:186
00267758.005 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0xe5136ad8/sipSPIInitiateDisconnect: Initiate call disconnect(38) for outgoing call
00267758.006 |17:15:19.364 |AppInfo |//SIP/SIPHandler/ccbId=4212/scbId=0/ccsip_api_call_disconnected: ccb->cc_disc_cause (38); ccb->sip_disc_cause(503)
00267794.001 |17:15:19.367 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 122.129.75.131 on port 23600 index 689
[7527,NET]
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 192.168.1.60:54441;branch=z9hG4bK00005d94;received=122.129.75.131
From: "42001" <sip:[email protected]>;tag=3c77e6734e8e00180000789e-000014f2
To: <sip:[email protected]>;tag=4211~132b515c-f941-ca80-8023-9423185ada2d-30582939
Date: Tue, 21 Apr 2015 12:15:14 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Server: Cisco-CUCM10.5
Reason: Q.850; cause=41
Content-Length: 0
Seems like a network issue. TCP connection can't be established with the called jabber client, which as per the trace is 192.168.1.172.
Check for firewalls blocking the connection.
You may take a pcap on the called jabber client side or probably from the call manager, and check for TCP issues.
You mentioned the jabber client's IP as 192.168.1.60 but I see incoming invite from 122.129.75.131. What this IP?
HTH,
Atul

Free Source Softphones Problem

Hi,
I'm trying to configure various free source softphones (linphone, ekiga and jitsi) to a cucm, but all of them failed. My question is, how can i configure an account to make them work properly.
While trying to see what fails in wireshark, it says that it can't find the account.
Thanks.

Hi Mate,
I had tried jitsi in past while testing something. I was able to register it to CUCM.
1) Created End user
2) Setup Digest credentials on the end user page(used by SIP client)
3) Created a basic third part SIP device in CUCM with a dummy mac address.
4) Assigned extension etc. and did setup DIGEST USER on the phone config to the user created in step 1
5) Associate the device to end user
from jitsi:
1) Add account> SIP ID: [email protected] (where 1.1.1.1 was my primary subscriber not Pub)
2) Pwd: Digest Credentials set above
3) Advanced>Connection>Authorization name: userid of the end user created in step1
This is what had worked for me
-Terry

T61 in docking w/ Avaya softphone slows MS Office apps

I have a T61 in type 2504 docking station. With Avaya RO6.01.81 softphone.
MS Office 2003 products execute slowly (1-2 minutes) w/ softphone plugged in any usb port on dock sta
Problem non existent when plugged dir in laptop.
Updated laptop BIOS from 1.26 to 2.24--no change
Removed all other apps except MS Office--no change
Disabled parallel port--no change
Updated MS Office--no change
Installed new Lenovo USB drivers--no change
Updated R06 Sp5--no change
Cleaned terminals on laptop and dock station--no change
The source of the problem appears to be in the docking station since no problem occurs when plugged in laptop

so what's up with the docking station? Is there a firmware fix for this? We use a lot of these 2504's in this company and they all behave the same way. Is there firmware on the dock? Users forget to plug the softphone in their laptops in the morn and miss their calls. Anybody?

BT Softphone and D-Link 2740B router problem

Has anyone managed to work out how to set up a D-Link 2740B router so that BT Softphone will work?
The best help I can get from BT INDIA is change the baud to 5060, what ever that means?
I am not a techie, so any help in a simple step by step fashion would be most welcome.
Solved!
Go to Solution.

no thats no what i thinking about... i dont want to connect any devices as handsets ect to my laptop, laptop must be free as is now if this router have VOIP in the lastest firmware i think i can connect router to my bt broadband line then standard handset to the router and thats it. From the router level set the voip softphone internet telephony connection for this handset and thats it.... then if I will need another standard landline handset then simply connect it straight to the bt line... thats my vision....no devices connected to my computer its not working that way
If you already have bt home hub its working away from your computer so only one reason to change router is to do it in the same way as BT.... as BT have rubbish routers and with more than one laptop in the home network just cut the connecton this making me mad.....

WRT54G does not work or what am I doing wrong?

My setup: DSL -> 2wire modem router (bridge mode, firewall shutdown) ->WRT54G -> LAN with 1 IPPBX, 1 wired PC and 3 wireless PC.
VoIP is running fine within our LAN, we have 2 SIP trunks running fine, But we just cant make an external softphone to work properly from WAN (outside the LAN).
As long as I know, the simplest way to explain what a firewall does is 'everything requested from the inside should be wellcomed, anything unsolicited from the outside should be disallowed UNLESS explicitly told otherwise' this is, why the heck do we have unreachable ports althought we have allow those to receive extern traffic?
Please check the pdf´s (snapshots) here:
http://travelexpress.com.mx/download/router/status.pdf
http://travelexpress.com.mx/download/router/router_firewall.pdf
http://travelexpress.com.mx/download/router/router_forward.pdf
http://travelexpress.com.mx/download/router/scanner.pdf
http://travelexpress.com.mx/download/router/scanner1.pdf
All those are for showing current setup, firewall, port forwarding and the results of a port scanner wich shows that efectively, we cannot use our extern SIP client because the WRT54G is blocking it...
Thanks in advance as we are urged to fix this ASAP

Yes. NAT is a function in the router. Just like the firewall. But still, both are different functions of the router. The firewall filters traffic to the router. NAT is responsible for mapping IP addresses. The reason why unsolicited packets from the internet won't get into the LAN is because NAT cannot deliver those packets and not the firewall rejecting some incoming connection. NAT does not filter. NAT tries to deliver and if it does not know where it drops it. But not because it wants to filter something.
The settings on the router are correct. But as I have mentioned before, forwarding a port does not mean it's open or accessible. There are still other factors which could cause the port to be closed...
Regarding the modem: you can check that if you connect a computer directly to the modem, open a port on the computer and then do the internet port scan again.
I also explained how to check with telnet if TCP port 5060 on the IPPBX is really open or not. Why don't you just do this test? It would be much easier and faster, giving relevant answers. I don't mind whether asterisk usually uses both protocols on 5060 or not. Maybe it does. Maybe not. Maybe it can be configured. Whatever. Enter "telnet 192.168.1.69 5060" into a command prompt window of a computer connected to your LAN and you see if the port is open or not. Then we know something for sure instead of thinking about what should be or should not be...
Also do the same telnet test with port 23 and 80 and then use the internet port scanner to scan for tcp port 23 and tcp port 80. Then we could see if anything goes through or not instead of discussing theory...
I did not say you have to set up a lab setup. I only explained that you were mixing NAT functions and firewall functions of the router. The NAT function may look like some firewall in certain respects but it is still a different function with a completely different purpose.
I also can't say if the IP settings are all correct. Maybe the IP address is correct but the gateway address is not. Or the DNS server.
The subject of this thread is "... or what am I doing wrong?". It's impossible to comment on something if you don't give the details which would allow to verify that you have set it up indeed correct...

CUPC client not showing softphone but showing up under presence viewer on CUP server

Hello,
Having a strange issue where the presence client wont show the softphone configured for them but when you go to the presence server and use the presence viewer to see what the client should be seeing, it shows up there. From that it looks like the UPC phone in call manager is being seen ok by the CUPS server but for some reason not by the client. Tried fully shutting down the client and also restarting the re-sync service on CUPS but no such luck.
thanks

Please check if TFTP is configured on CUPS (CUPS > Application > Settings)
Also trying pulling the softphone manually from CUPC client workstation
Command (from windows command prompt)
tftp -i GET .cnf.xml
If this doesn't successfully downloads the file there is firewall issue in the network or on the workstation over tftp.
(Note: Windows 7 doesn't have tftp client installed by default, need to install it from Control Panel > Programs and Features > Add/Remove Windows Feature)

WHAT IS A SOFTPHONE ??

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