@RobinRiekkola8641 wrote: Ok i will ask them to open up several ports for the RTP packets and try again, i will review the admin manual, didnt have that available from the beginning. Thank you Good. Hopefully that will solve the problem. Of course if you do not open the entire range configured on the SPA112 you will need to use the "Phone Adapter Configuration Utility" built into the SPA122 to configure the range you have enabled. Go to Voice>SIP>RTP Port Min and RTP Port Max   Probably not necessary but if you continue to have a problem and wish to capture a sip debug trace, the SPA112 Phone Adapter Configuration Utility settings are as follows: 1. Voice>Line1> Sip Debug Option: FULL 2. Voice>System>Debug Server>[ipaddress of syslog program] 3. Voice>System>Debug Level>1 You need a syslog program and I previously posted an address to obtain a simple windows syslog program. ... View more

I have asked the IT department to open up port 5060 TCP for the SIP protocol and port 16384 UDP for the RTP protocol in the firewall, between my ATA and the SIP server they have handed me.   The SPA112 uses a range of port numbers for RTP packets. The admin manual says you should define a range that contains at least 4 even number ports, and if you don't change anything the default port number range is 16384 to 16482 (Admin Guide pg 68). I believe calls only use a single port number for a given call however the adapter doesn't necessarily use the 1st port number in the range.   ... View more

Reviewing the problem: The objective of using the SPA112 Analog Terminal Adapter is to enable the connection of the intercom system's output to a distant cellphone connected to one of the cellular networks. Basic Review: Cabling: The intercom system's analog output is designed to connect to a telephone network and dial a number? Using a cable with an standard RJ11 connector you have connected the intercom system's output to the SPA112's "Phone 1 or Phone 2" port. Is this correct? The SPA112's "Internet" port is cabled to a distant router that attaches to a local network and the internet. Correct? The SPA112 is cabled to a local power source. Configuration: To connect to a telephone number in a cellular network you need to go thru a sip server that can pass the call to the cellular network. This would be a voip service or a PBX server that has a connection to a voip service. What kind of service are you connecting to? Is it located at a local IP address or an external IP address? You have configured Line 1 or Line 2 Tab (depending on the cabling from the intercom system). You indicated that you set Make Call Without Reg: Yes The Sip Server accepts unregistered calls? Is this correct? On the Line Tab you have configured Proxy: (Domain Name or URL) User ID: xxxxxxx Password: xxxxxxxx Is this correct? If you are not having the SPA112 Register with the Sip Server, I would set Register: No You indicated the you set the Sip Transport to TCP. This should be OK. The RTP packets are still sent as UDP. Your router should not be blocking UDP. For testing I would think instead of attaching the intercom system to the SPA, you could just attach an analog phone and get that working first, then later go to the intercom system. To debug your problem it would be helpful to see a Sip Debug Trace. The trace will show the SIP Invite you send when you make a call and will show the response (if any) from the sip server. You can get this with a computer attached to the same network as the SPA112. You install a Syslog Server Program on the computer and configure the SPA to send the trace to the computer. On the SPA you put the computer's address and activate the Logging. See the Cisco SPA100 Series Phone Adapters Administration Guide for detail on the settings. On the SPA112 You make settings on the Line Tab (Sip Debug Option: FULL) and under Administration > Log >Log Server under System>Debug Server the ip address of your computer and also under System>Debug Level set 1   You can download a simple syslog server program that runs under Windows here ... View more

You haven't described your use of the SPA112 in enough detail to understand your setup, however I can make a comment about "one-way audio" that you mentioned.   As part of the sip protocol for making calls the SPA tells the distant server the ip address to which the distant server should send the incoming audio rtp packets. Depending on the environment the ip address can be a local ip address or an external internet ip address. One cause of one-way audio is the protocol's failure to send the correct ip address. Often this is corrected by the NAT Mapping Enable setting. If the SPA needs its external ip address and has discovery problems there are additional settings. ... View more

The YeaLink SIP-T46G is a modern IP Business phone. I assume you have verified that you can make a direct unregistered IP call. 1002@IPAddress:5061 in a speed dial should work IPAddress can be either a local network address or an external ip address. If it is an external ip address then you may get into router blocking problems because the call is not solicited. Blocking problems are solved by port forwarding. I would try to verify that the sip INVITE command is physically received by the SPA112. The SPA has a logging function that may show the incoming sip detail. If you have a capability to capture the network traffic with WireShark it would be even better. On the SPA112 Line 2 should have the following settings Line Enabled Sip Port 5061 UserID: 1002 Ans Call Without Reg: Yes Make Call Without Reg: Yes (if you want to make outgoing calls) If you are receiving a call over the external ip address you probably need NAT Mapping Enable: Yes   ... View more

The SPA112 Quick Start Guide says the device has an option that you can enable or disable to "Allow or prevent WAN access to the administration web server". The manual says the menu option is 7932.  Using IVR accessed from the phone attached to the SPA112 you should make sure this option is enabled. If it is disabled you would not be able to access the SPA for configuration as you are attempting. You can download the Quick Start Guide here    ... View more

The "hotline", where you pickup the phone and it automatically dials a predetermined number, is done with the SPA122's dial plan. You change the default dial plan to a special one the accomplishes your requirement. They are described in the SPA100 Series Phone Adapters Administration Guide. https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag_1_3_5.pdf See page 145 The Admin Guide suggests  [code] (P0<:1000>) [/code] where the example is dialing 1000 ... View more

Paul, Thanks for checking out the phone line. From the trace it looked like was should be working OK but I couldn't think of an additional problem.   A copy of your configuration could help give me new ideas. If you can't attach the configuration that you saved with your browser then I don't know how to send it on this forum. I can't deal with your .pdf copy. The .pdf copy was a printed copy of the configuration's .html code.   As an alternative, email the old fashioned copy that you saved with your browser to your hard drive to my temporary email address: hwharbor-cisco@yahoo.com   A couple of quesions: When you make the failed call do you hear anything at all? What is the voltage level of the PSTN line when it is idle, on-hook? Howard ... View more

Paul, Are you confident you have no problem with the PSTN line? Can you attach a phone directly to the cable attaching to the FXO port and make calls to your numbers?   The SPA3102 INFO screen you posted says the SPA3102 is still connected to the PSTN line, still off hook, but your text says it was after the call. The PSTN line voltage should be about 48v when the line is idle, not 7v and off hook as shown on the screen.   I compared your trace to a trace of a similiar call on my SPA3102 and my trace looked pretty similiar to yours. The sip trace shows you are trying to call 150 on the attached PSTN line. I believe the trace shows that it connected to the FXO port successfully but apparently you heard no sound. Neither trace actually showed the dial digits going to the off hook PSTN line but apparently that is not something the trace shows. The traces seem to indicate that sound is being transferred by rtp and the info screen indicates rtp bytes were sent and received.   A key setting on the PSTN Line Tab is One Stage Dialing: Yes,  however I think you have that set correctly or you would get a dial tone.   ... View more

The SPA3102 PSTN Line Tab is designed to allow interfacing a legacy PSTN telephone circuit with voice over internet circuit. The PSTN Line Tab is more or less a separate voip adapter inside the SPA3102. It allows a PSTN caller to send his call to its destination over voip. Conversely it allows a voip caller to terminate his call on the public switched telephone network.   VoIP registration is a process that tells a voip server, a voip provider or PBX, the ip address and port number of a sip client that can receive a phone call and keep that server updated if the address changes.   You can setup a voip account on the PSTN Line Tab and register to a voip server and send incoming calls to the voip-to-pstn gateway or receive calls from the pstn-to-voip gateway and send them to a voip server for termination.   With the PSTN Line Tab incoming voip calls can only be routed to the voip-to-pstn gateway FXO port. There is no provision to allow the call to go out to another voip client like a softphone.   My impression of what you are doing with the softphone and the SPA3102 PSTN Line Tab is not the standard use of the SPA3102 PSTN Line capabilities. I believe you are using the PSTN Line Tab to allow a computer softphone to make and receive phone calls using your PSTN Line. I believe you are interfacing the SPA3102 with the softphone using direct ip addressing.   I would look for a softphone that allows you to configure multiple accounts. Setup one account for the direct ip calling and another to directly register with your voip provider to receive the incoming calls from your DID phone number. I am confident that you can find something to meet your needs. ... View more

Then you need to setup your softphone to register to an account at your voip provider and route incoming calls to your DID directly to your softphone.   You can either setup a 2d account or subaccount for the softphone or setup the SPA3102 to Register: No.   I would setup Line 1 of the SPA3102 to not register (Register No) and to Make and Receive Calls without Reg. Many/most voip providers will allow you to make calls without registration.   The Cfwd All on the User Tab doesn't work for you because the SPA3102 itself doesn't do the forwarding, the SPA3102 sends a Sip command to your voip provider that there is a change in destination and it is up to the voip provider to send the call to the new destination. Most voip providers will not honor that request due, I believe, to fraud considerations. An Asterisk PBX and other Sip clients generally will honor the request.   The no dialtone is probably due to your setting of Make Calls Without Reg: No. In this case with this setting the SPA3102 does not give a dial tone when the SPA3102  is not registered to the voip provider. ... View more

If I understand you correctly you are asking if you could switch an incoming call to an existing softphone instead of using the phone attached to the SPA3102.   You don't go into much detail on the current softphone accessibility except to say that the softphone can communicate with the SPA3102 PSTN Line Tab and you say it uses the same network. I do not believe the SPA3102 was designed to internally allow routing a call from the Line 1 Tab to the PSTN Line Tab voip, only to the PSTN Tab FXO port.   If the softphone is registered to a voip server you may be able to do an attended transfer of the incoming call directly to the softphone.   If the softphone is not registered to a voip server, then if you want to try something that is non-standard, since the softphone is communicating with the PSTN Line Tab you may be able to separately communicate with it using its local network ip address and port number. This, of course, will depend on the softphone capabilities. I would first see if I could dial the softphone from the phone attached to the SPA3102 using a speed dial. In other words setup one of the Speed Dial Settings on the User 1 Tab with the softphone ip_address:port_number. In other words, 1777123123@168.1.104:7901 where 1777123123 is the softphone userid, 192.168.1.104 is the local network ip address and 7901 is the softphone port number. You may have to do some detective work to figure out the port number which hopefully doesn't change over time. Then see if you could dial the speed dial number from the phone attached to the SPA3102. If this is successful then you may be able to do an attended transfer of the incoming call by putting the incoming call on hold with a flash and dialing the speed dial number when the second dial tone comes up.   I roughly checked this out on my SPA3102 with an incoming call switching it to an X-Lite softphone on my computer. Depending on other settings, there may be some audio problems related to NAT settings on the SPA3102 that you will need to deal with. ... View more