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Audio on Account 2 but not on Account 1


#1

Technical Information

Sangoma Linix 7 (Core) (x86_64)
Kernel Version: 3.10.0-862.9.1.e17.x86_64
FreePBX Version: 14.0.5.25
Asterisk Version: 15.5.0

Background

Installation is in the cloud and is serving two (2) locations, about 20 minutes apart. Agency1 has 10 phones, Agency2 has 4 phones. Of those, 11 user phones are Sangoma S705 and 3 conference room phones are Sangoma S500.

Three users are alternating between the two locations. As such, they each have an S705 phone at each location and I have their phones configured with two accounts so that they can make/receive calls for either agency. This way their outgoing calls have caller ID for the appropriate agency. Inbound Routes with different Alert Info (distinctive ring) handles incoming calls.

UserA and UserB are primary to Agency1. UserC is primary to Agency2. As such, the phones are configured as follows: (Note the difference in phone number assignments for UserC)

Agency 1 - UserA
Extension 122 uses Account1 and Agency1 phone number 111-111-1111
Extension 322 uses Account2 and Agency2 phone number 222-222-2222

Agency 1 - UserB
Extension 133 uses Account1 and Agency1 phone number 111-111-1111
Extension 333 uses Account2 and Agency2 phone number 222-222-2222

Agency1 - UserC
Extension 125 uses Account1 and Agency2 phone number 222-222-2222
Extension 325 uses Account2 and Agency1 phone number 111-111-1111


Agency 2 - UserA
Extension 222 uses Account1 and Agency1 phone number 111-111-1111
Extension 422 uses Account2 and Agency2 phone number 222-222-2222

Agency 2 - UserB
Extension 233 uses Account1 and Agency1 phone number 111-111-1111
Extension 433 uses Account2 and Agency2 phone number 222-222-2222

Agency2 - UserC
Extension 225 uses Account1 and Agency2 phone number 222-222-2222
Extension 425 uses Account2 and Agency1 phone number 111-111-1111

Issue

I provided a lot of info to help you understand my dilemma. Everything has been working well until two days ago when the phone lines for Agency1 were inadvertently disconnected (thanx, AT&T). They were restored later that day. However, there were issues with those six phones.

Initially they were not registering. That was corrected. All phones at Agency1 are functioning properly. Two phones at Agency2 are functioning properly (UserB and the conference room). The quirky issue is with Agency2, UserA and UserC.

Both accounts on the phones for UserA and UserC are registered. However, calls made or received using Account1 have no audio. Calls made and received using Account2 are fine. Phones ring appropriately for incoming calls but, when answered, there is no audio if the phone number is using Account1. This is true for both incoming and outgoing calls whether the phone uses the speaker or the handset. Note that the phone numbers for UserA and UserC are reversed so it does not seem to be a phone number issue. And UserB is working fine.

Any ideas?


(Dave Burgess) #2

One-way audio is usually a NAT problem with the firewall/router. There a ton of possibilities, including the FreePBX server at location 2 needing to be restarted to restart the IP address resolution.

A SIP debug (syntax varies based on channel driver) should give you all the information you need to start troubleshooting.


#3

Thanx for the reply Dave

Audio silence is both ways. No changes have been made to the router / firewall. No servers at either location. One server set up in the cloud handles both locations. The only phone equipment at the locations is the phones themselves.

As additional information, the AT&T phone numbers are currently forwarded to numbers at a sip trunk provider. And as mentioned, it’s been working for a couple of months. The process is underway to port the numbers to the sip trunk provider but not sure that will have any effect.

Got me stumped.


(Dave Burgess) #4

OK, sounds like it’s time for some SIP debug. Simply coming back and saying “Nope” (which is how I interpret your answer) doesn’t move the ball forward.

The NAT thing is still our number one suspect. In fact, unless you’ve got some kind of Codec mismatch (which would cause the call to fail on pickup), there are almost no things that can cause audio problems that aren’t provider/router/firewall related.

So, I’m going to say it again.

One-way audio (even when it’s both ways) is either going to be a NAT issue, an external addressing problem, or a router/firewall config. Remember that FreePBX has a firewall too, so you need to make sure that all of the addresses in there are correct as well.

I recommend using the pastebin link provided by Sangoma (see Consistent Asterisk/FreePBX Crash Issue) and post the “full” log output from a failed call with SIP DEBUG turned on, perhaps one of us will be able to spot the problem.


#5

Sorry Dave. Didn’t mean for it to come across that way. Just clarifying the installation. I appreciate your help. Turned off SIP ALG in the router and it seems to have corrected the issue.