FreePBX Distro 3.211.63-10 Paging Issues

I am finding out that under the current Distro that Paging seems to be blowing up badly for us under the current distro. We setup a paging group in FreePBX, it picked extension 777 by default, and proceeded to try and test it.

When we try and page, we get the following errors spitting out of Asterisk:

[2013-06-22 10:49:03] WARNING[13085][C-00000159]: chan_sip.c:21863 function_sippeer: 
SIPPEER(): usage of ':' to separate arguments 
is deprecated.  Please use ',' instead.

[2013-06-22 10:49:03] WARNING[12807][C-00000096]: 
channel.c:1309 __ast_queue_frame: Exceptionally 
long voice queue length queuing to 
Local/[email protected]ing-0000004a;1

I get many of each error in the logs, and then finally at the end I get:

/usr/sbin/safe_asterisk: line 159: 11105 Segmentation 
fault      (core dumped) nice -n $PRIORITY
 ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} 
> /dev/${TTY} 2>&1 < /dev/${TTY}
Asterisk ended with exit status 139
Automatically restarting Asterisk.

Needless to say this is a BAD thing, and leaves piles of Aastra SIP phones in a limbo state.

Anyone have any ideas, or is paging busted in the current FPBX release. This is a major issue for this install for sure…

It’s definitely not ‘busted’ by any means. Worked fine on my installed and test base two months ago and nothing has changed since then in the code base.

You havent stated if you are using meetme or app_conference

I am using app_conference in the configuration.

Are the warnings about the argument separator to be expected, as I don’t see how I could have introduced this one?

It actually did page the other day, when only about half the phones were online, now with pretty much everything online it seems to be broken. Any chance I am running into an issue due to the large number of extensions in the group and system? As this was the “page all” group…

Thanks for the response…

How many extensions are we talking about?

Oh about 250 give or take a few… It will grow from that point up to something in the range of 350-400 in the end.

On the strange side, I am going to drag my partner over with me tomorrow, as I want to see what he was doing when it crashed (outside of paging), as I went there earlier, and I was paging fine, it takes it a moment to open the speaker on that many phones, but it did work… I think I still see that separator error, but it didn’t seem to crash.

I will work with this more during the day, and see if it keeps running, crashes, or if it decides to be a total you know what and just do it random…

What type(s) of phones are you running? You really should be doing multicast paging.

You are talking almost 20M of traffic to signal and page all those phones. Double that up for the responses. I frankly am surprised that it worked at all.

The good news is multicast works very well, the audio is concurrent to all stations (although multiple stations in an area that are within each others acoustic aperture are going to have some distortion/cancellation artifacts due to phase cancellation/comb filtering).

Humm, learn me something new as they say, would you mind sharing more info on doing this multicast paging with Asterisk/FreePBX, as it sounds like you have worked on it. We still need to catch up on the phone some day, I know we keep saying we will do that.

As to the phones, they are all Aastra 6737i’s with the exception of a few Soundstation IP 7000’s in the conference rooms. Bandwidth wise it’s not an issue,as I just built the entire network for them as well during this move, so we have a batch of Cat6500’s doing power and network distribution around the place, with GigE to every phone/workstation, and 20Gig channels (using 10GE) between the switches, so even 50meg of traffic is just a non-issue network wise.

Still using multicast for the in-building phones sounds interesting, and I will dig a little on it, but would for sure be interested in hearing more about it from you if you care to share…

-Howard

The Multicast has nothing to do with FreePBX at all. The phones “listen” on a port and anything that comes on that port goes out the speaker (yes, you can play games and stream music out the default address or, if a customer just isn’t listening to you play Homer Simpson’s “can you dig it” over the whole office, just don’t plan on getting invited back.

I would have to look up in my cheat sheet but all you do is program a button as a multicast originator. You can break into groups/sub-groups too.

BTW Aastra’s hate loads of extraneous multicast traffic as they have to sort through it. Make sure you have multicast flood control on your 6500 interfaces.

Thanks for the tip, I have a page all multicast group up and running to about 250 phones, and it seems to work well. Granted I’d still be interested in any notes/recommendations you have on your cheat sheet when you have your hands on it.

As to the Cisco cats, what type of multicast configuration do you find works well. I don’t think anything else is doing any multicast, and all of the phones/pbx are all in their own VLAN, so there really shouldn’t be any other multicast traffic moving around…

It doesn’t matter the phones are in their own VLAN if the port is in trunk mode and the data VLAN is enabled on the trunk port (to use the PC port).

The Aastra software will freeze and show no service. Also causes voice issues.

Microsoft Sharepoint use to be a big culprit of this.

Good point for sure, and yep they do use the PC ports on the phones, one of the reasons for the 6737i’s was the fact that it had two GE ports on the phone.

I have dropped you a private message as well…