I have a SPA3102 Voice Gateway with Router, as voip technologies are mainly migrating to PJSIP protocol, I’m facing SPA3102 gateway problems to route calls from VOIP to PSTN.
By the way I have tried all possible variants on configuration and all test call get response with the message 503 service unavailable message.
Therefore I asked to TAC about this issues, with the reply to my inquiry; no longer support since Nov 30, 2019, suggesting post a question regarding this issue to Cisco community for assistance.
If anyone has any information about it, I will appreciate it.
Unfortunately, we don't know whats wrong with call setup. Requirements vary between providers. You need to know why call gets rejected with 503 response. Ask support of the operator rejecting it. Or at least disclose SIP requests and responses captured on wire ...
Again I get in touch with “tac” support before, with the reference to VOIP to PSTN calls with 503 message response, as I mentioned on this post.
Unfortunately with the solely response about the product discontinuity and suggesting to post this question to Cisco community.
Regarding capture, I have worked with ATA debug option and syslog, which could be different with wireshark? if both way will get 503 service unavailable message, without know causing reason?
By the way according to ietf documentation
503 Service Unavailable The server is temporarily unable to process the request due to a temporary maintenance of the server. The server MAY indicate when the client should retry the request in a Retry-After header field. If no Retry-After is given, the client MUST act as if it had received a 500 (Server Internal Error) response. A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD attempt to forward the request to an alternate server. It SHOULD NOT forward any other requests to that server for the duration specified in the Retry-After header field. The client SHOULD block traffic to a server based on the servers IP address and not the hostname since hostnames can represent multiple servers. Servers SHOULD NOT refuse the connection or drop the request as a replacement for responding with 503 (Service Unavailable).
Clearly this refers to inside the ATA gateway software and nothing could be done without Cisco support.
Sorry, I misunderstood the call direction. It's SPA3102 returning the 503 error, not remote VoIP proxy (e.g. incoming, not the outgoing call). In such case I need logs from SPA3102 as well as SIP packets captured.
Finally could know why SPA3102 response 503 message.
It becomes from FXO port line polarity issue, So if on configuration screen “voice -> info -> PSTN Line Status) the option “line voltage” with negative voltage value, SPA3102 will response in any try call from VOIP to PSTN with 503 unavailable message.
It could sound or look illogical but is true, it seems FXO port detect line polarity, crazy moreover when is regardless to loopback PSTN interface anyone device on market it have design with line polarity detection.
By the way on product specification and other related document is not mentioned anything (about line polarity either installation and configuration testing).
So it have nothing to do with (PJ)SIP ...
POTS line has never been polarity agnostics. Yes, very old analog phones doesn't care, most modern digital phones can do automatic polarity switching, but you should not bet blindly on it. Some analog system are using polarity reversal as signal before CID transmission. It's better to honor the polarity during wiring ...
I guess the negative voltage has been considered another signal here. If line voltage decrease bellow threshold (and negative voltage is bellow), the line is considered "in use" (by other phone device connected in parallel). In-use line is unavailable for calling.
Nice to see "POTS" and "standard" without "no" in the same sentence.;-)
As far as I know, there is nothing like "standard" for analog lines in public telephony. There are per-country standards or even per-company standards. While they are similar to each other, they vary in almost all parameters.
DC voltage, ringing voltage, ringing wave shape, dialing style (I remember pulse dialing in my country, which has used different pulse rate than pulse dialing in Italy), impedance, different methods of seizure detection, ...
There is no standard even for wire names. In my country we have just A and B wire (A is negative).
I know branch PBX operating on 24V only (and some "standard phones" can't work with it). I know phones assuming sine-wave ringing tone that can't cope with trapezoid wave shape. I'm operating PBX branches in seven countries around - and despite all of them are near European countries, the "standards" vary. I meet so many incompatibilities between "standards".
I'm not trying to overpower you in dispute. Just be careful assuming there is "standard" for POTS line. True - you can buy random phone (POTS terminal) and there's very high chance it will work with particular public (and even branch) network, but it may not work and sometime it doesn't work because "standards" vary so much. It needs to be taken into consideration if there's a issue (like in your case).
The SPA3102 has a configuration setting on the PSTN line Tab: "Line-In-Use Voltage"
My Admin manual says:
"Determines the voltage threshold at which the SPA-3000
assumes the PSTN is in use by another handset sharing the
same line (and will declare PSTN gateway service not
available to incoming VoIP callers)."
I recall cases where this setting must be changed (lowered) in an environment where the analog line is operating at less than 48v. Generally it is set about half way between on-hook and off-hook voltage. Polarity is not important.
Have you tried lowering this setting?
Glad you got the SPA working.
My experience is that PSTN lines show 48v when on-hook and about 6v or less when off-hook. You can read the voltage on the INFO Tab. As Dan Lukes said, the SPA is also used with some PBX systems where the on-hook voltage is only 24v. The SPA "Line-in-Usse" setting is a setting on the SPA to tell the SPA software the voltage level to consider the line in use.
Hi Howard Wittenberg, thanks again!
My conclusion, this issue it become due to LINE-In voltage threshold
value and way software handler define, more when it related to negative
line voltage values (I can't understand why this device it detect
polarity for what??), it seems by modifying LINE-In value it make works.
Can’t believe there is no information in troubleshooting reference,
could make a references about 503 message and possible causes information.
In conclusion I have tested by connecting to PBX extension port and
voice communication is acceptable, but when it connected to PSTN network
the echo becomes intolerable (despite adjusting gain levels values but
voice communication hearing low with lower levels or with echo with
About you mentioned off-hook voltage level (6 volts when go to off-hook)
it can can raise level by increasing value of "Operational Loop Current
So my final conclusion is:
In comparison with other ATAS, the experience SPA 3102 it unsatisfied
(moreover when I though Cisco VoIP products have a wide experience).
Anyway thanks again for your contribution. I hope this could be useful
I share capture screen of INFO Tab
The INFO tab shows that your PBX line voltage (31v) is lower than the SPA3102 Default Line-In-Use setting. As a result on an incoming voip call which you have setup to forward to the PSTN (FXO) port the SPA3102 software felt the line was already in use and it would not allow you to forward your voip call to your local PBX extension. You lowered the Line-In-Use setting to a value below the PBX on-hook line voltage and you were able to forward the call.
Your echo problem is more complex. A google search turns up this Cisco document that discusses reducing SPA3102-PSTN line echo and may have some workaround solutions.