Still new to all of this but I setup a freepbx system. I have been unsuccessful at making any outbound calls. Flowroute has told me that my Content Length sent to them is incorrect and I’[m not sure where to start looking for the problem. Below is their response and I’ve also attached the log from the call.
I’ve located the invite,
it appears that our switch is chocking on the invite you are sending because of this line.
Content-Length: 3100
This Content-Length is not accurate to what you are sending in the SDP.
The actual content length in the invite is 290 not 3100.
The log does not have SIP trace info, so I’m just guessing that the problem is a SIP ALG in your router/firewall. Try turning that off. Any setting that mentions SIP is suspect. If you are double NATted (router is connected to a modem/gateway that also does NAT), put the modem in bridge mode if possible, or check it for any SIP related settings.
If you still have trouble, at the Asterisk command prompt, type pjsip set logger on
make a failing call, paste the new log (which will now include a SIP trace) at pastebin.freepbx.org and post the link here.
Also, please post: ISP? Modem make/model? Separate router/firewall make/model, if any?
Did you download raspbx-10-10-2020 and then run raspbx-upgrade? If not, explain how you built the system.
I don’t know whether this is an issue for you, but my Flowroute account requires sending the tech prefix (8 digit account number followed by *) along with the destination number. If this is required on all accounts, you’ll have to set that up once you get past the present problem.
Thank you so much for your quick reply. I genuinely appreciate it.
The log does not have SIP trace info, so I’m just guessing that the problem is a SIP ALG in your router/firewall. Try turning that off. Any setting that mentions SIP is suspect. If you are double NATted (router is connected to a modem/gateway that also does NAT), put the modem in bridge mode if possible, or check it for any SIP related settings.
My modem/router is one single unit. NAT ALG SIP was enabled. I toggled it off and now call attempts reported to my softphone are “forbidden,” rather than, “number not available,” so it does appear to have been a problem.
If you still have trouble, at the Asterisk command prompt, type pjsip set logger on
make a failing call, paste the new log (which will now include a SIP trace) at pastebin.freepbx.org and post the link here.
Also, please post: ISP? Modem make/model? Separate router/firewall make/model, if any?
Did you download raspbx-10-10-2020 and then run raspbx-upgrade? If not, explain how you built the system.
ISP: Suddenlink Communications
Modem/Route Combo: Older Motorola Surfboard
I downloaded raspbx-04-04-2018 and did raspbx-upgrade. It’s an older version to support a raspberry pi w with a usb-to-ethernet adapter.
I don’t know whether this is an issue for you, but my Flowroute account requires sending the tech prefix (8 digit account number followed by *) along with the destination number. If this is required on all accounts, you’ll have to set that up once you get past the present problem.
This is an intriguing bit of info that customer service did not mention to me during our chats.
My next steps after your advice
Flashed the device with freshcopy of raspbx-04-04-2018. The setup with the basic details for sip service provided by flowroute knowledge base. I remain getting the forbidden error. The log above is from this call.
There is unfortunately still no SIP trace. To get that, at the Asterisk command prompt (not a shell prompt), type pjsip set logger on
and the response should be PJSIP Logging enabled
Also, note that Apply Config (or anything that restarts or reloads Asterisk) turns pjsip logger back off, so you should make the test call right after turning logger on.
I believe that Flowroute is fussy about the From domain. If you don’t already have it, set From Domain for the trunk to sip.flowroute.com
According to
the Tech Prefix is needed only for IP authentication, though you might try it anyhow:
In settings for your Trunk, under Dialed Number Manipulation Rules, set Outbound Dial Prefix to your Tech Prefix followed by *. On my account, the prefix is the same as the user name, so you could try
setting Outbound Dial Prefix to 60193407*
If no luck, take that out and make a new test call with pjsip logger properly enabled, paste the log and post the link.
There is unfortunately still no SIP trace. To get that, at the Asterisk command prompt (not a shell prompt), type pjsip set logger on
and the response should be PJSIP Logging enabled
Also, note that Apply Config (or anything that restarts or reloads Asterisk) turns pjsip logger back off, so you should make the test call right after turning logger on.
I previously was using SSH to get into the PBX/asterisks and rebooted every-time I killed the connection. I apologize, I thought it continuously logged it rather than turning off.
I believe that Flowroute is fussy about the From domain. If you don’t already have it, set From Domain for the trunk to sip.flowroute.com
This was previously blank but I’ve tried it today with and without this detail
According to ### This article describes the steps required to prepend your Flowroute Tech Prefix to dialed numbers in order to use IP-Based Authentication (https://support.flowroute.com/173409-Set-Up-IP-based-Authenti the Tech Prefix is needed only for IP authentication, though you might try it anyhow:
In settings for your Trunk, under Dialed Number Manipulation Rules, set Outbound Dial Prefix to your Tech Prefix followed by *. On my account, the prefix is the same as the user name, so you could try
setting Outbound Dial Prefix to 60193407*
If no luck, take that out and make a new test call with pjsip logger properly enabled, paste the log and post the link.
Since yesterday, the freepbx dashboard noted some modules out of date. (likely overlooked from the fresh install) I updated all modules and rather than my softphone showing forbidden, now I receive the recording the all circuits are busy and have manually end the call.
The From header is still showing your local IP. Confirm that your trunk has From Domain set to sip.flowroute.com
I can’t make sense of the Destination Blacklist error on line 434. Confirm that in the Flowroute portal, you have SIP Credentials enabled for outbound.
The From header is still showing your local IP. Confirm that your trunk has From Domain set to sip.flowroute.com
adjusted back to that in this log
I can’t make sense of the Destination Blacklist error on line 434. Confirm that in the Flowroute portal, you have SIP Credentials enabled for outbound.
Flowroute support just responded to my email and said the following:
Hi THOMAS,
I think i’ve found the right call. The caller id is one digit off from what you have listed. The account you are using to make calls is set up in a way in the fraud control not not allow ANY calls to complete.
Possibly, on a new account, they are fussy about outbound caller ID. The 817-444 number you are sending (if it wasn’t redacted) seems strange; it was assigned to Frontier and never ported. Try using your Flowroute DID instead.
All issues that you folks have helped me work through were active issues.
The final 403 blacklist issue was related to flowroute defaulting to a strict whitelist of countries. None of which were the US and I was dialing US numbers.
One more thing: In Asterisk SIP Settings, External Address and Local Networks are not correctly set. This didn’t cause trouble with Flowroute, because they detect it and do NAT traversal on their end. But for most providers or if you want any external extensions, these settings are important.
