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Buddy List Firewall Block

Posted by IVSCOMM on Mon, 05/03/2010

I have a customer with a fancy firewall (ISA) and the buddy list is not working with polycom phones.

It works great outside of the office but not in the office.

Any ideas?

Is it a port block issue?

If so what port is that information being sent out on?


Submitted by IVSCOMM on Tue, 05/04/2010 Permalink

Just the BLF. Although I would like to know how to populate the list automatically...

However, I have manually entered the names and the status (wheteher or not they are on the phone) is not being shown.

If I take the phones outside of this network it works perfectly.

By the way the phones work perfectly in every other way.

Submitted by IVSCOMM on Thu, 05/06/2010 Permalink

Ok so it's not a firewall issue. Here's the scoop

setup
Ext.200 (ip650) set up to monitor ext.400 (ip430)
Ext.400 (ip430) set up to monitor ext.200 (ip650)

Test 1
If i call out on the 430 the 650 shows that the extension is busy (the circle with a dash in it)
If i put the caller on hold on the 430 the 650 shows that the extension has a caller on hold (the clock symbol)
Once I take the caller off hold it goes back to the busy sign.
If i hang up from the caller the 430 the 650 shows that i am now available to take a call again (the waiting person)

Everything is as it should be..right? wrong!

Now if I reverse everything this is what happens...

Test 2
If i call out on the 650 the 430 shows that the extension is available to take a call (the waiting person)
If i put the caller on hold on the 650 the 430 shows that the extension has a caller on hold (the clock symbol)
Once I take the caller off hold it goes back to the waiting person NOT busy.
If i hang up from the caller the 430 the 650 still shows that i am available to take a call (the waiting person)

I have placed a packet sniffer on the 430 and it does not receive the sip packets for busy only for on hold. but it does get the packets if I am monitoring any other model of phone that I have as long as it is not a 650 even if its an aastra phone it works as prescribed.

xxxx
xxxx Bang head here!
xxxx

Shawn

Submitted by eeman on Thu, 05/06/2010 Permalink

but your sure its not the firewall? you said both these devices work perfectly somewhere else. What if its a NAT issue, and the 430 was the first device to come online and register and get PAT set up with the firewall, and the 650 was second.

try taking them both off line for a while, long enough for the firewall to forget about them (overnight?)
then bring up the 430 first, then after its up bring up the 650... if the same behaviour reverse this order on 2nd test and confirm same behaviour again.

Submitted by IVSCOMM on Thu, 05/06/2010 Permalink

I was wrong in my initial statement. I had only tested with a 430 and a 650 and only watching the 430 on the 650 not the other way round. I verified this behavior on site with the customer and it is doing the exact same thing. only the 430 works properly not the 650.

Shawn

Submitted by IVSCOMM on Fri, 05/07/2010 Permalink

I have checked it a bazillion times. The funny thing is when you put a call on hold that shows but not when you are on the phone.

I can see the packet come across to signal the light but there is no packet sent for the actual call only when you put the call on hold.

Submitted by eeman on Fri, 05/07/2010 Permalink

so does the sip extensions setting for the 430 have a call limit set? for the notifications to work it has to count the calls. perhaps one extension has call-limit but the other does not?

Submitted by IVSCOMM on Thu, 05/13/2010 Permalink

Well not really. It's the same thing. the 650's don't work on my system. I worked with Mike White from e4 technologies and he used one of his 650 phones on my system to test and it came up the same way (yes, he had the buddy list turned on) He had the latest firmware, so I am positive it is not a polycom issue. We checked the sip.conf files and everything looks ok (or so he told me I don't know what I should be looking for). Here is the File

[330-TenantName]
qualify=yes
nat=yes
accountcode=TenantName
pickupgroup=2
;=model=polycom-601/650/670
callerid=Polycom <330>
context=from-inside-TenantName
;=mac=MacID
canreinvite=yes
vmexten=330
secret=XXXXXX
host=dynamic
username=330-TenantName
subscribecontext=local-extensions-TenantName
callgroup=2
dtmfmode=rfc2833
type=friend
mailbox=330@default-TenantName
disallow=all
allow=ulaw
allow=gsm

I will be checking to see if the phone is actually sending the sip signal and will let you know soon.

Does anyone have any suggestions?!

Shawn

Submitted by eeman on Thu, 05/13/2010 Permalink

when you say 'he has the latest firmware' did you somehow provision his phone of some other server and not your pbx? because regardless of the firmware that was on the phone, they always grab the version where the config files are to ensure compatibility. So if he ftp'd your config files, he also revised his sip.ld to match whats on your server.

Submitted by moshe on Tue, 06/22/2010 Permalink

i recently realized (not sure if it started recently or i just realized it) that some extensions cannot be monitored i have switch phone for these extensions to rule out a phone issue put behind other networks to rule out firewall or other network issues and still the same outcome when i check buddies it shows as away, i have that with a few of my customers now, i cannot figure out the issue and don't know where to start trouble shooting

any thoughts

Submitted by eeman on Wed, 06/23/2010 Permalink

FYI call-limit must be set in order for asterisk to keep track of call state. The previous example given does not have call-limit set on the extension even though a previous thread of mine clearly says it must be set.

Submitted by moshe on Mon, 07/05/2010 Permalink

just got this issue again (the extension have call limit set on it) i did core show hints and the result was 61555@local-extensions-TEN: SIP/61555-TENANTNAME State:Hold Watchers 2

than i did a show channels i see that extension 61555 is holding a active channel however there is no call on the line it seems like a call is stuck running
SIP/MTE1-0a746e10 (None) Up AppDial((Outgoing Line))
SIP/61555-TENANTNAMEdial-SIP@macro-tl-di Up Dial(SIP/TENANTNAME+1845791300

this is a call i made on Friday where the phone was unplugged in the middle of a conversation luckily its a internal call so Im not being charged for it
A few questions
1. how could i force a channel hangup?
2. how could i make sure this is not happening with external calls, like if a call is silent for a 60 minutes it should disconnect ( if it all possible)

any insight would be of help thanks