I have set up an internal extension system using a Grandstream HT813 and RasPBX (FreePBX). The office’s analog telephone line is connected to the FXO port of the HT813. The system is functioning well—internal extensions work flawlessly, and incoming calls are successfully received on mobile apps like Zoiper and PortSIP installed on the users’ phones.
However, I’m encountering an issue when making outbound calls from these apps (Zoiper, PortSIP, etc.). When dialing an external number through the Outbound Routes following the configured dial pattern, I hear a continuous tone, similar to the one heard when picking up a traditional analog phone (a dial-tone). At this point, I need to open the app’s dial pad and manually input the number again to successfully connect the call.
As far as I understand, the HT813 is supposed to automatically transmit the dialed number to the external line without requiring manual input.
To troubleshoot, I have adjusted the “Wait for Dial-Tone” and “Stage Method” settings under the Channel Dialing section of the HT813 in various combinations, but the issue persists.
Could you suggest any steps or settings I might try to resolve this problem?
I suggested something simple that matches numbers mentioned by the OP of that thread. You don’t have to change your dialplan. Just paste a log and we’ll go from there.
Thank you for the clarification.
I enabled logging and made a test call as suggested. Here are some notable logs that stood out:
[2024-12-20 04:58:47] ERROR[26131]: res_pjsip_header_funcs.c:410 remove_header: No headers had been previously added to this session.
[2024-12-20 04:58:49] WARNING[26131]: res_pjsip_pubsub.c:3345 pubsub_on_rx_publish_request: No registered publish handler for event presence from 101
Do you think these logs are related to the issue, or should I look for something else? Let me know if you need additional details.
Your paste shows that the called number is missing from the INVITE sent to the HT. But we can’t see why, because you pasted console output, not the Asterisk log, which can be found in /var/log/asterisk/full .
Paste the section created by your test call. Leave the paste expiration at the default (never) so future visitors to the thread can follow along.
The Outbound Route is a problem, though I don’t know whether it is the only problem.
As a test, in the route To-HT813, remove the prefix (leave it blank) and put X.
(X followed by a dot) in the match pattern.
Dial a number in the same format as you would on an analog phone connected to the PSTN line.
If the call fails, paste the Asterisk log (not the console output), including pjsip logger.
If the call succeeds, tell us how you want the patterns to work. Give an example of what you would dial and what should be sent on the PSTN line.
I’m following this topic too. @Stewart1, yes, it also works for me with the dialplan set this way " X."
As @jungeon mentioned, I also have to wait about 10 seconds before the outgoing call is actually initiated, but only on the analog line (FXS) connected to the HT813. I don’t have this problem with VoIP phones.
I tried reducing the numbers as you suggested in my other topic, but it seems they have no effect. I wonder if this is normal.
For me it is no big deal, just a curiosity, but perhaps for others the wait might be problematic.
Sorry, I stand corrected, even for my VoIPs the wait is several seconds, between all the steps, although a little faster than FXS.
My logic is that it is usually 3 seconds per call. In this case, 2 calls are made, one to the gateway and the other to the outgoing number.
Maybe that’s why it comes to 6/7 seconds.
But that’s my guess. I don’t know if there is a way to adjust these times.
.