SOLVED Alertinfo and Paging Issue

flyingtodd

New Member
Joined
Jun 27, 2011
Messages
3
Reaction score
0
I'm an asterisk nu b and I am having a heck of a time getting my Polycom IP500's to auto answer a page.

I'm running PIAF 2.9 w/ Incredible PBX that I just installed a couple of days ago.

The page group is set up and I believe the sip.cfg files are proper with Alertinfo settings. (see this thread http://pbxinaflash.com/community/threads/polycom-scripts.7299/?t=7299)

When ever I dial 801 (page group) all of the phones just ring. If you answer a ringing phone manually you can then here the page. Also when I dial *80ext# it too just rings.

When you initiate a page from an ext the phone "beeps" like it should to prompt you to start talking. In the meantime all the phones ring twice and then stop ringing. The ext that initiated the page does not hang up until you press "end call"

I am also using the endpoint manager for the polycom's.

Any tips would be appreciated as I have scoured the web for insight, but everything I do doesn't help.

Thank you in advance,

Todd
 
Last edited by a moderator:

tbrummell

Guru
Joined
Jan 8, 2011
Messages
591
Reaction score
54
The problem is in the config file for the Polycom. I've dealt with this before. I'm in a meeting right now, but when I'm done I'll try and locate the information. It has to do with the AlertInfo setting, so you're on the right track!
 

flyingtodd

New Member
Joined
Jun 27, 2011
Messages
3
Reaction score
0
Thanks for the tip...I'll post my progress

Well, I haven't figured it out yet, but I am getting closer. I installed TB 2.6 on another machine, just to see if worked there, and it did right out of the box. So I will reference some of the cfg info from there to see if I can figure it out on PIAF 2.9
 

flyingtodd

New Member
Joined
Jun 27, 2011
Messages
3
Reaction score
0
Solved

I figured it out :smile5:

My system is PIAF 2.0.6.2 & Asterisk 1.8.8.0 with the Endpoint Manager configuring my Polycom 500's, 501 and 430.

The EPM creates all of the config files for the phones, but the file(s) I was overlooking were the server_213.cfg, server_317.cfg and the server_325.cfg. Within those files they create the following:

<alertInfo
voIpProt.SIP.alertInfo.1.value=""
voIpProt.SIP.alertInfo.1.class=""/>

From the reading I have done, value should be Ring Answer and the class should be 4. I simply removed those lines rebooted the phones and the server and paging/intercom worked like a charm.
 

swf2e

New Member
Joined
May 24, 2013
Messages
3
Reaction score
0
Location
Murfreesboro, TN
I am having the same issues as described in this thread. I have completed the modification listed in the solved post above and still are not able to get paging/intercom to work in my environment. The phones are Polycom 501 and 601 using the 3.1.8 software on them.

This is the last thing in order to finish the project up. If anyone has any ideas or suggestions then I would appreciate the help. This is the first time I have setup a phone system of any kind from scratch.

The last time I dealt with them was several years ago (the links in FreePBX were along the left of the windows instead of across the top in a menu, and we didn't use paging at all. There were three of us that used that system, the server was in a datacenter in Atlanta and we were scattered across Tennessee.

PBX in a Flash PURPLE Status Program
┌───────────────────SYSTEM INFORMATION *VERIFIED*─────────────────────┐
│ Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE │
│ Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PIAF Installed Version = 2.0.6.4 under *HARDWARE* │
│ FreePBX Version = 2.11.0.1 │
│ Running Asterisk Version = 1.8.22.0 │
│ Asterisk Source Version = 1.8.22.0 │
│ Dahdi Source Version = 2.6.2 │
│ Libpri Source Version = 1.4.12 │
│ IP Address = XXX.XXX.XXX.XXX on eth0 │
│ Operating System = CentOS release 6.4 (Final) <> │
│ Kernel Version = 2.6.32-358.2.1.el6.i686 - 32 Bit │
└─────────────────────────────────────────────────────────────────────┘

NOTE: FreePBX 2.11 is installed, but the results were the same under version 2.10.
 

moodinsk

New Member
Joined
Oct 12, 2011
Messages
14
Reaction score
1
What isn't working? What does the asterisk console say when you attempt an intercom/page?

Are you using OSS Endpoint Manager?

Like FlyingTodd mentioned, you need to remove the garbage alertinfo from server_318.cfg

In sip_318.cfg, make sure that your alertinfo is set to:

Code:
<alertInfo voIpProt.SIP.alertInfo.1.value="Auto Answer" voIpProt.SIP.alertInfo.1.class="3"/>
 

swf2e

New Member
Joined
May 24, 2013
Messages
3
Reaction score
0
Location
Murfreesboro, TN
What isn't working? What does the asterisk console say when you attempt an intercom/page?

Are you using OSS Endpoint Manager?

Like FlyingTodd mentioned, you need to remove the garbage alertinfo from server_318.cfg

In sip_318.cfg, make sure that your alertinfo is set to:

Code:
<alertInfo voIpProt.SIP.alertInfo.1.value="Auto Answer" voIpProt.SIP.alertInfo.1.class="3"/>
I apologize for being so vague in the previous post. Like I said, I am new to these systems and that was my first post.

The problem was as described in the first post above. The phones will ring twice and then disconnect. The person trying to page out hears a single tone and then it looks to them like it is working. Trying a page directly to a phone *80XXX will do they same, except the person initiating the call will be disconnected after the two rings.

I am using OSS Endpoint Manager to deploy the configuration. The two changes (removing lines from server and adding to sip) have been completed as suggested.

What do I need to look for in the console? Is there something I need to copy from the system for people to look at? If so, where is that information saved? I will have to wait until I can get back on-site to do any testing.
 

moodinsk

New Member
Joined
Oct 12, 2011
Messages
14
Reaction score
1
Your best bet is to SSH to your PBX and go into the asterisk console: asterisk -r

Attempt and intercom and copy the output here. My guess is that you might still a rogue AlertInfo tag somewhere in your templates.
 

swf2e

New Member
Joined
May 24, 2013
Messages
3
Reaction score
0
Location
Murfreesboro, TN
I don't know what changed, but something somewhere is allowing it to work now.

Thank you for your help. If something breaks again, I will be on here again.
 

nunya

Member
Joined
Oct 1, 2010
Messages
44
Reaction score
3
Since some of the links seem to be broken... I had a customer whose paging quit working after updating to FPBX 2.10. Same thing as the OP, all phones ring twice. All Polycom phones.
Found the solution here at the very bottom: freepbx.org/trac/ticket/6052
 

dudbug

New Member
Joined
Sep 25, 2013
Messages
11
Reaction score
1
Nunya,
Great post! I have tried several changes and I can't seem to get this to work. I have tried both Ring Answer and Auto Answer in the sip.cfg file. I have also tried creating a simple entry into ext-local that looks like this:

exten => *33,1,SIPAddHeader(Alert-Info: Auto Answer)
exten => *33,2,Dial(SIP/1003)
exten => *33,3,Hangup

It is almost like the Polycom 501 isn't respecting the Alert Info data. It seems odd that the Wireshark shows the alert info data as info=Ring Answer. I have even modified the extensions_additional.conf and changed this so that it only comes in as Ring Answer (I also tried Auto Answer)

I did change the settings in MySql for this as well but they don't seem to be used.

Either way...any more suggestions are welcome.

1. OOS manager used
2. Modified Mysql paging (no help)
3. Tried modifiying alert info in SIP.cfg (didn't help either)
4. Changed settings in extensions_additional.conf to try to make Polycom 501 happy (nope..still no worky)
5. Checked for updated firmware for said Polycom (I'm already at 318)
6. Proceeded to pull remaining hair out (complete)

Please find attached my config file..in anyone can take mercy on me and help, that would be appreciated.


Here is a failed attempt.

-- Executing [*[email protected]:1] Goto("SIP/1001-00000000", "ext-intercom,*801003,1") in new stack
-- Goto (ext-intercom,*801003,1)
-- Executing [*[email protected]:1] Macro("SIP/1001-00000000", "user-callerid,") in new stack
-- Executing [[email protected]:1] Set("SIP/1001-00000000", "AMPUSER=1001") in new stack
-- Executing [[email protected]:2] GotoIf("SIP/1001-00000000", "0?report") in new stack
-- Executing [[email protected]:3] ExecIf("SIP/1001-00000000", "1?Set(REALCALLERIDNUM=1001)") in new stack
-- Executing [[email protected]:4] Set("SIP/1001-00000000", "AMPUSER=1001") in new stack
-- Executing [[email protected]:5] Set("SIP/1001-00000000", "AMPUSERCIDNAME=1001") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/1001-00000000", "0?report") in new stack
-- Executing [[email protected]:7] Set("SIP/1001-00000000", "AMPUSERCID=1001") in new stack
-- Executing [[email protected]:8] Set("SIP/1001-00000000", "CALLERID(all)="1001" <1001>") in new stack
-- Executing [[email protected]:9] GotoIf("SIP/1001-00000000", "0?limit") in new stack
-- Executing [[email protected]:10] ExecIf("SIP/1001-00000000", "0?Set(GROUP(concurrency_limit)=1001)") in new stack
-- Executing [[email protected]:11] GosubIf("SIP/1001-00000000", "7?sub-ccss,s,1(ext-intercom,*801003)") in new stack
-- Executing [[email protected]:1] ExecIf("SIP/1001-00000000", "0?Return()") in new stack
-- Executing [[email protected]:2] Set("SIP/1001-00000000", "CCSS_SETUP=TRUE") in new stack
-- Executing [[email protected]:3] GosubIf("SIP/1001-00000000", "0?monitor_config,1(ext-intercom,*801003):monitor_default,1(ext-intercom,*801003)") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/1001-00000000", "0?is_exten") in new stack
-- Executing [[email protected]:2] StackPop("SIP/1001-00000000", "") in new stack
-- Executing [[email protected]:3] Return("SIP/1001-00000000", "FALSE") in new stack
-- Executing [[email protected]:12] ExecIf("SIP/1001-00000000", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [[email protected]:13] GotoIf("SIP/1001-00000000", "0?continue") in new stack
-- Executing [[email protected]:14] Set("SIP/1001-00000000", "__TTL=64") in new stack
-- Executing [[email protected]:15] GotoIf("SIP/1001-00000000", "1?continue") in new stack
-- Goto (macro-user-callerid,s,26)
-- Executing [[email protected]:26] Set("SIP/1001-00000000", "CALLERID(number)=1001") in new stack
-- Executing [[email protected]:27] Set("SIP/1001-00000000", "CALLERID(name)=1001") in new stack
-- Executing [[email protected]:28] Set("SIP/1001-00000000", "CHANNEL(language)=en") in new stack
-- Executing [*[email protected]:2] Set("SIP/1001-00000000", "dialnumber=1003") in new stack
-- Executing [*[email protected]:3] Set("SIP/1001-00000000", "INTERCOM_CALL=TRUE") in new stack
-- Executing [*[email protected]:4] GotoIf("SIP/1001-00000000", "0?end") in new stack
-- Executing [*[email protected]:5] GotoIf("SIP/1001-00000000", "0?end") in new stack
-- Executing [*[email protected]:6] GotoIf("SIP/1001-00000000", "0?allow") in new stack
-- Executing [*[email protected]:7] GotoIf("SIP/1001-00000000", "0?nointercom") in new stack
-- Executing [*[email protected]:8] GotoIf("SIP/1001-00000000", "0?nointercom") in new stack
-- Executing [*[email protected]:9] Set("SIP/1001-00000000", "DEVICES=1003") in new stack
-- Executing [*[email protected]:10] GotoIf("SIP/1001-00000000", "0?end") in new stack
-- Executing [*[email protected]:11] Set("SIP/1001-00000000", "LOOPCNT=1") in new stack
-- Executing [*[email protected]:12] Set("SIP/1001-00000000", "_SIPURI=") in new stack
-- Executing [*[email protected]:13] Set("SIP/1001-00000000", "_ALERTINFO=Alert-Info: Ring Answer") in new stack
-- Executing [*[email protected]:14] Set("SIP/1001-00000000", "_CALLINFO=Call-Info: <uri>;answer-after=0") in new stack
-- Executing [*[email protected]:15] Set("SIP/1001-00000000", "_SIPURI=intercom=true") in new stack
-- Executing [*[email protected]:16] Set("SIP/1001-00000000", "_DOPTIONS=A(beep)") in new stack
-- Executing [*[email protected]:17] Set("SIP/1001-00000000", "_DTIME=5") in new stack
-- Executing [*[email protected]:18] Set("SIP/1001-00000000", "_ANSWERMACRO=") in new stack
-- Executing [*[email protected]:19] GotoIf("SIP/1001-00000000", "0?pagemode") in new stack
-- Executing [*[email protected]:20] Macro("SIP/1001-00000000", "autoanswer,1003") in new stack
-- Executing [[email protected]:1] Set("SIP/1001-00000000", "DIAL=SIP/1003") in new stack
-- Executing [[email protected]:2] ExecIf("SIP/1001-00000000", "0?Set(DIAL=DAHDI/1003)") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/1001-00000000", "0?macro") in new stack
-- Executing [[email protected]:4] Set("SIP/1001-00000000", "phone=") in new stack
-- Executing [[email protected]:5] ExecIf("SIP/1001-00000000", "0?Set(CALLINFO=Call-Info: <sip:broadworks.net>;answer-after=0)") in new stack
-- Executing [[email protected]:6] ExecIf("SIP/1001-00000000", "0?Set(ALERTINFO=Alert-Info: Intercom)") in new stack
-- Executing [[email protected]:7] ExecIf("SIP/1001-00000000", "0?Set(ALERTINFO=Alert-Info: Intercom)") in new stack
-- Executing [[email protected]:8] ExecIf("SIP/1001-00000000", "0?Set(ALERTINFO=Alert-Info: info=Auto Answer)") in new stack
-- Executing [*[email protected]:21] ChanIsAvail("SIP/1001-00000000", "SIP/1003,s") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*[email protected]:22] GotoIf("SIP/1001-00000000", "0?end") in new stack
-- Executing [*[email protected]:23] GotoIf("SIP/1001-00000000", "0?godial") in new stack
-- Executing [*[email protected]:24] Set("SIP/1001-00000000", "CONNECTEDLINE(name,i)=1003") in new stack
-- Executing [*[email protected]:25] Set("SIP/1001-00000000", "CONNECTEDLINE(num)=1003") in new stack
-- Executing [*[email protected]:26] Dial("SIP/1001-00000000", "SIP/1003,5,IA(beep)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/1003
-- Connected line update to SIP/1001-00000000 prevented.
-- SIP/1003-00000002 is ringing
-- Nobody picked up in 5000 ms
-- Executing [*[email protected]:27] ExecIf("SIP/1001-00000000", "?Return()") in new stack
-- Executing [*[email protected]:28] Busy("SIP/1001-00000000", "20") in new stack
== Spawn extension (ext-intercom, *801003, 28) exited non-zero on 'SIP/1001-00000000'
 

Attachments

nunya

Member
Joined
Oct 1, 2010
Messages
44
Reaction score
3
I'm trying to remember... I've slept a few times since then. The Paging issue is pure Polycom. They told the FreePBX paging module developers to change the alertinfo tag to something other than the old “ring answer” in the SIP header. Polycom has a newer software version out that is not retro-compatible with the 501. They failed to take this into consideration, or consider 501 to be past their “end-of-life”. This had to be modified in the database – Use phpMyAdmin.
I think the ultimate solution was to change "info=Auto Answer" to "Ring Answer".
It's late. Don't hold me to it.
 

dudbug

New Member
Joined
Sep 25, 2013
Messages
11
Reaction score
1
This seems to be pretty well documented if you stitch the 40 or so articles that talk about this. They all refer to changing the alert info to something that the phone will understand and work with in the cfg file. The database change that you refer to is in mysql under asterisk paging somthing or other. The problem is that PIAF doesn't seem to use the database for this functionality. When viewing wireshark, the only place that I could make changes was in extensions_additional.conf. I tried just about every combo that I could think of to get these confounded phones to respond....My VVX500s work perfectly without tinkering.
 

PBXEHR

Member
Joined
Sep 30, 2013
Messages
42
Reaction score
0
When I try paging from a Polycom 670 asterisk -r gives me: WARNING[27329][C-000001bd]: chan_sip.c:22045 function_sippeer: SIPPEER(): usage of ':' to separate arguments has been deprecated, use ',' instead. ERROR[27329][C-000001be]: pbx.c:4390 ast_func_write: Function PJSIP_HEADER not registered
 

kenn10

Guru-ish
Joined
Dec 16, 2007
Messages
845
Reaction score
148
I gave up on PJSIP. Too many issues. Paging headers work fine with chan_sip.
 

PBXEHR

Member
Joined
Sep 30, 2013
Messages
42
Reaction score
0
Hi kenn10, What do I change on the Polycom phone to make it use chan_sip instead of PJSIP? I've gone into the Polycom Phone Web Browser and turned on the softkey for paging.
 

kenn10

Guru-ish
Joined
Dec 16, 2007
Messages
845
Reaction score
148
On the extension screen, change it to chan_sip from PJSIP on the drop down menu and change the port number to match 5060 or 5061, whichever is set in your advanced settings. You also need to change the Advanced setting under PJSIP and SIP to insure you do not have the same port numbers for both.
 

Members online

PIAF 5 - Powered by 3CX

Forum statistics

Threads
22,260
Messages
136,393
Members
14,499
Latest member
BenMcClements