(crossposted)
Hi everyone, I am configuring a fxo gateway (Vega 50) to work with FreePBX/Asterisk, for the most part it works with the exception that caller ID works half of time, sometimes works and sometimes I get a “Anonymous” caller ID.
At this point I am asking if somebody knows what fine tuning needs to be setup so the fxo gateway can see the caller ID all time, the POTS provider is Telnor/Telmex in Mexico.
Here the fxo settings I have currently:
Here the tones setup:
thanks!
sometimes works and sometimes I get a “Anonymous” caller ID.
What exactly means??
“Anonymous” could be possible because calling party telephone line is not figured a public line (well knows as “Private” line and on any originated call caller ID presentations is set as restricted) or could set before making a call to set temporally caller ID presentations as restricted, better should contact your provider, about caller ID service.
OP has made multiple postings on multiple forums, regarding this. One needs to see From: "Anonymous" <sip:Anonymous@x.x.x.x>.... why? - Sangoma - FreePBX Community Forums for the background to this, and Who knows how to configure POTS gateway parameters for Mexico Telmex/Telnor? (caller id works intermittently) - Endpoints - FreePBX Community Forums for the multi-posting.
Thank you for bringing this to my attention. I apologize if my actions have caused any inconvenience or violated any forum guidelines. I understand the importance of keeping discussions organized and avoiding duplicate posts.
I posted the question in the FreePBX forum after encountering the issue with Caller ID intermittently not working, hoping to gather insights from a broader community. However, I see now that it may have been perceived as redundant.
I assure you that I value the community guidelines and will make sure to adhere to them in the future. If there’s anything specific I can do to rectify the situation, please let me know, and I’ll address it promptly.
Thank you for your understanding and assistance.
-Aldo
Moving it to FreePBX was right, but you should have cross-linked the threads, so that people know the full history. I think, for example, the suggestion that the call was anonymous wouldn’t have been made if they had seen that you made all the calls.
I totally agree.
Thanks!
Whatever, this is a misunderstanding of Caller-ID services (Data communication, message, etc.).
As mentioned it get “Anonymous”, it means caller ID information is being received (just in case of no message received it could be a data corruption or no caller ID FSK signal have received).
Isn’t easy to debug about FSK caller ID signal even if could capture FSK data it needs special device to debug, only can connect an external Caller ID device and make a comparison.
The OP claims that caller ID works when the FXO gateway is replaced by an analogue phone. I don’t think they have tried the two in parallel, and failures, in that case, could be attributed to the parallel connection.
The OP claims that caller ID works when the FXO gateway is replaced by an analogue phone.
Well that point isn’t figured on the post (…encountering the issue with Caller ID intermittently not working…), with Vega gateway.
could be attributed to the parallel connection.
Technical and electrical parallel connection, it’s improbable could causes intermittently, moreover old PBX systems which not supporting caller id a topically solution is connect in parallel a caller Id device.
That is correct, I am making all the testing with two cellphones that have caller id enabled, and if I use an analog phone connected directly to the pots line, caller id 100% of time
When I do the same testing but with the sangoma vega 50 as fxo gateway, it works only 50% of time (more or less).
At this point I am not sure what else to check, are there some known parameters of analog lines that must check or that should tweak? Sangoma fxo gateway can see the caller id, it just don’t process it all times or can’t catch it all times I think if we can call it that way.
At this point I am not sure what else to try, do you guys think its a hardware or firmware issue with sangoma vega? or it just need some adjustments… (I am using the last firmware available for download dated on 2018)
And ultimately, should I test with different hardware? maybe another brand of fxo gateway? if so what would be the most proven to work brand? I choose sangoma vega because supposedly reputable but even paid sangoma support didn’t accept my equipment serial number I think because they don’t support this model anymore.
Aldo
Well, not so much can said about it.
There are two ways of Caller ID detection mode (data transmission during ringing or data transmission prior to ringing), the most common is the first one. Data transmission shall occur during the first long silent period between first ring and second ring patterns.
Can’t said technically exactly what happening, if gateway is correctly interpreting the caller ID signal detection sequence (sequence between the first RP-AS Ringing Pulse Alerting Signal ring where the caller ID FSK signal is transmitted).
Could be possible to connect only IP phone with Vega?? have a CLI
command debug?
Do you think some operators (carriers) use a mix of during ringing and data transmission prior to ringing methods? or is either one or the other? I ask because when vega detects the caller id it does immediately after I dial the pots number, and when the same time that I did setup on tones.def.5.off_time=“4000” (4 seconds) occur before the first ring is when vega does not detect or not display the caller ID and uses the parameter: advanced.sip.anonymous_display_name=“Anonymous” that according to the documentation that parameter is used when vega cant detect a caller ID or when the caller ID is actually not present or private.
If sometimes works and sometimes don’t what can it means?
I didn’t understand that, so my current setup is POTS → vega → asterisk → pjsip extension, and the connection between vega and asterisk is an analog trunk.
Can please define what you mean by “only ip phone with vega”? And by the way, yes vega have a CLI command debug.
Thanks
Aldo
I would say that indicates it is using an after first ring method. However, I would expect it to use different methods for different caller ID type settings. Unfortunately, the manual I found doesn’t list the options. I’ve now found one that does list the options, and GR30 is is the standard, US, after first ring method.
I wouldn’t expect this on the same line. With the before first ring methods, there is generally a signal before ringing is applied, and a slight delay before ringing.
In FreePBX speak, the connection between Vega and Asterisk is a SIP trunk!
I believe he means use an IP phone, rather than Asterisk, to receive the call from Vega, although I think that is clutching at straws.
Have you asked the person who sold you the gateway?
Do you think some operators (carriers) use a mix of during ringing and data transmission prior to ringing methods?
A. No, FSK modulation is transmitted between first ring and second (First ring signal → CALLER ID FSK Modulation → second ring → third ring, etc.).
I ask because when vega detects the caller id it does immediately after I dial the pots number,…
and when the same time that I did setup on tones.def.5.off_time=“4000” (4 seconds) occur before the first ring is when vega does not detect or not display the caller ID and uses the parameter: advanced.sip.anonymous_display_name=“Anonymous” that according to the documentation that parameter is used when vega cant detect a caller ID or when the caller ID is actually not present or private.
A. Sorry this is related to VEGA, but I don’t think so “ on tones.def.5.off_time” parameter have relationship with caller ID feature (better to look someone with VEGA experience).
advanced.sip.anonymous_display_name=“Anonymous”, that’s for meet SIP Headers requirements, when no CID information SIP headers Caller ID information wil be “Anonymous”.
Could be possible to connect only IP phone with Vega?? have a CLI
Some gateway brands allow to use as leased line PSTN<–>(FXO GW) — IP Network — (FXS GW) Phone.
Good luck.
That depends on the system in use.
BT, in the UK, sends tones before the first ring. It is possible that some other UK operators use a mix of the BT and after ring system, but if they do, it won’t be on the same line. Originally the other operators tended to use US market equipment.
The BT system does a battery reversal to signal the start of the call, followed by an audio tone. Terminals that support their system apply a partial, temporary, loop to the line, starting with a short. full loop. The advantage is that you can implement services that don’t disturb the user by ringing.
(SIN 242 is a bit clearer about the timing of the first ring.)
Mexico seems to use the US system, so is a an after first ring system.
Practically all countries use ETSI FSK and Bellcore except UK uses V23 FSK prior first ring signal.
not really, he is just an ebay warehouse that sells overstock, not specialists. I tried to contact official sangoma support but they took me down after found out my device serial number eol.
I was reading in some forums that my carrier uses Bellcore for caller id.
These are the caller ID methods supported by vega 50:
In this vega documentation says this:
gr30-sdmf
Conforms to Bellcore standard GR30 - single data message format. Just passes the call time and
number information. The latest standard mentions that this format may be dropped in future.
gr30-mdmf
Conforms to Bellcore standard GR30 - multiple data message format. This passes the caller name
as well as the call time and number. (This configuration will also receive gr30-sdmf caller Id
So that confuses me about bellcore, again because the default for USA/Mexico is bell202, is Bellcore standard GR30 the same thing as bell202 ? or like totally different formats?
I wonder why if it does not support specifically bellcore, when I use etsi-fsk, etsi-fsk-lr, or etsi-fsk-post, caller id works 50% of the time (more or less), that makes me wonder if my carrier uses etsi-fsk and vega cant interpret it all times, or in fact my carrier uses bellcore but sometimes is able to get the caller id when I choose etsi-fsk… i have no idea!
I googled other supposedly reputable big players fxo gateway brands that are not too expensive and I found this cisco that just went eos literally some days ago: SPA 8800, it supports the following caller id types (Bellcore is one of them):
I just purchased and I will post the results of my testing once I receive, install, configure, and test the new cisco unit.
Note: I am so desperate that I am even using chatgpt for this issue, and this is what chatgpt told me about the
GR-30 SDMF (Single Data Message Format) is a signaling protocol standard defined by Telcordia (formerly Bellcore) for transmitting Caller ID information. It’s commonly used in North America, including Mexico, for Caller ID transmission over analog telephone lines.
So yes, the GR-30 SDMF codec should work for receiving Caller ID in Mexico. This codec is specifically designed to decode the Caller ID information transmitted using the SDMF format, which is the format typically used in Mexico and other North American countries.
And when I asked about my specific scenario it gave me sort of generic answer but mentioned couple of things that called my attention:
If Caller ID is working inconsistently when using the GR-30 SDMF codec in your FXO gateway in Mexico, there are several possible reasons for this:
1. Carrier Support: Ensure that your carrier supports Caller ID and is transmitting Caller ID information reliably for all incoming calls. In some cases, carriers may have issues with their Caller ID infrastructure, leading to intermittent or incomplete transmission of Caller ID data.
2. Signal Quality: Poor signal quality or interference on the telephone line connected to your FXO gateway can result in Caller ID information being corrupted or lost. Ensure that the FXO gateway is connected to a clean and stable telephone line.
3. Configuration: Verify that the FXO gateway is configured correctly to handle Caller ID information. Check the configuration settings related to Caller ID detection, decoding, and timing to ensure they are properly set according to the specifications of the GR-30 SDMF codec and the requirements of your carrier.
4. Timing and Synchronization: Caller ID information is typically sent between the first and second ring cycles of an incoming call. If the timing or synchronization settings of the FXO gateway are not optimized, it might miss or misinterpret the Caller ID data. Adjusting timing settings or ensuring proper synchronization can help improve Caller ID reception.
5. Hardware or Firmware Issues: In some cases, hardware issues with the FXO gateway or outdated firmware can affect Caller ID reception. Make sure that the FXO gateway is running the latest firmware version and that there are no hardware issues affecting its performance.
Chatgpt mentioned possible timming and synchronization issues or firware issues… do you guys think that can be a factor in my case? I dont see any setting in the vega 50 related to timming or sync with pots line.
Some other people have mentioned that in Countries like Mexico, Argentina, etc the carriers use ATA modems to convert their sip to analog line (which is my case) and then the cheap end-user modems sometimes don’t provide supposedly the enough voltage for the caller id, but again my analog test phone works just 100% of time on this setting… I am soooooo confused big time
Thanks all!
It is very difficult can someone could give you an answer or a solution.
As it involves many factors about this issue, starting from devices technical specifications plus PSTN line condition (attenuation, signal noise level, etc. Caller ID IC and internal circuit, etc.).
By trying with another device it could solve or not.
About “Some other people have mentioned that in Countries like Mexico, Argentina, etc the carriers use ATA modems to convert their sip to analog line (which is my case)…”
Does your case need to convert SIP trunk to analogue??? or should be analogue trunk to SIP? (it understand your case is, analogue to SIP).
In case of Argentina Telcos are leaving POTS lines, due to maintenance cost and rather old with a high cost to change to new one (and others problems as copper line heist), so they are moving to Optical transmission line.
