@Robert41794 wrote: Even if I call my number to ATA directly from my cell phone, when I press 5 on the ATA line, I do not hear it on my cell phone.   For what is worth, I ran a test on my SPA122 looking to see if pressing the 5 button on the called phone (attached to the SPA122) could be heard on my calling mobile phone. I tried changing various DTMF settings on both my SPA122 and voip provider (voip.ms) I could not get it to fail.  The problem could also be in the mobile phone leg of the call   ... View more

The configuration link you posted configures the SPA3102 to automatically dial a number thru a voip provider you would setup on the SPA3102 PSTN Line Tab. In other words automatically forwarding the call to the number you setup in the dial plan.   What do you want the SPA3102 to do? Do you want the SPA to give the caller a dial tone and then dial the number the caller enters? This is done with the dial plan you setup on the PSTN Line Tab.  A simple dial plan could be ([*x]x.) which allows the caller to dial any number combination in addition to *.   Think of the SPA3102 as two separate line configurations with Line 1 Tab the conventional line for a voip phone and the PSTN Line Tab for the use of the PSTN Line. You don't really use any configuration on the Line 1 Tab, however it could be a good idea to configure the Line 1 Tab to use an attached handset to register to an account on your Asterisk system if just to make a call and test configuration to that point.   For the PSTN Line Tab you would configure an account setup on  your asterisk system checking the following settings: Line Enable: Yes NAT Mapping Enable: (Yes/No) Proxy: Register: (Yes/No) DisplayName: User ID: Password:   The dial play setup as PSTN Caller Default DP will determine what happens to the incoming call. Generally you will use the Default configuration settings. For any detailed questions and explanation of the settings you should refer to the SPA3102 Administration Guide   There are settings to disable the ringing of the phone attached to the spa and also to delay the answer of an incoming PSTN Line call (to receive an incoming caller id). ... View more

  On the Regional tab under Miscellaneous did you try adjusting the "FXS Port Output Gain"? The default setting is -3. The Administration Guide says Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call Progress Tones and DTMF playback level are not affected by the FXS Port Output Gain parameter. The default is -3.   ... View more

This is just a guess, however, a way that an incoming call to your voip number could be diverted to your mobile number is thru a setting you made in your voip account for routing the call in case the SPA112 does not respond to the incoming call signalling, or is busy, or just doesn't answer. Many voip providers have this optional feature. In this case the problem could probably be with your fixed wireless broadband service. I would think your voip provider's call detail logs should document this diversion. If your voip provider does have this option you might want to disable it so the call could go to voicemail on your voip provider's system.   The one-way audio problem you mentioned frequently turns out to be a network address translation (NAT) problem involving the SPA112 and your external IP address. ... View more

@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