Quick Background:
I have quite a few PBX’s that are currently running Trixbox 2.6.2.2 and * 1.4 (bleh) and I am working on upgrading them all to take advantage of all the latest and greatest. Everything was going great until I hit this snag and I am hoping that someone here can point me in the right direction.
Current Setup:
┌────────────────────────System Information───────────────────────────┐
│ Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = OFFLINE │
│ Fail2ban = OFFLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ BlueTooth = OFFLINE | Hidd = OFFLINE | NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PBX in a Flash Version = 1.7.5.7 Running on *HARDWARE* │
│ FreePBX Version = 2.9.0.7 │
│ Running Asterisk Version = 1.6.2.20 │
│ Asterisk Source Version = 1.6.2.20 │
│ Dahdi Source Version = 2.5.0.2+2.5.0.2 │
│ Libpri Source Version = 1.4.12 │
│ Addons Source Version = 1.6.2.3 │
│ IP Address = 192.168.1.2 on eth0 │
│ Operating System = CentOS release 5.7 (Final) │
│ Kernel Version = 2.6.18-274.3.1.el5 - 32 Bit │
└─────────────────────────────────────────────────────────────────────┘My problem:
I have a phone environment that includes Polycom 550, 330, and 501s (bleh). Currently on trixbox, all phones are able to auto answer (page, intercom, etc) without issue. I fired up a test box and installed the newest FPBX distro and setup a basic environment for testing. While going through and testing to make sure all the features that I normally use were functioning, I noticed that whenever I tried to intercom/page my 501’s, they would ring a few times and then hang up. I figured that I just messed up my sip.cfg and forgot to include the Ring_Answer AlertInfo and all would be ok… Well, I verified that it is in fact there and being passed to the phone, it just will not auto answer for some reason. I tried PIAF (what I am currently running - see info above), and it has the same issue. I even tried with Asterisk 1.6 and still no go.
This has to be a configuration issue, right? The only difference between the two are the SIP versions. IP550/330 uses SIP 3.3 and IP501 uses SIP 3.1.7.
The 550’s & 330’s work just fine and auto answer as intended. What gives?
Intercom - IP330 -> IP501
AMG*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*80210@from-internal:1] Goto("SIP/211-00000014", "ext-intercom,*80210,1") in new stack
-- Goto (ext-intercom,*80210,1)
-- Executing [*80210@ext-intercom:1] Macro("SIP/211-00000014", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/211-00000014", "AMPUSER=211") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/211-00000014", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/211-00000014", "1?Set(REALCALLERIDNUM=211)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/211-00000014", "AMPUSER=211") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/211-00000014", "AMPUSERCIDNAME=330Test") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/211-00000014", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/211-00000014", "AMPUSERCID=211") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/211-00000014", "CALLERID(all)="330Test" <211>") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/211-00000014", "0?limit") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/211-00000014", "0?Set(GROUP(concurrency_limit)=211)") in new stack
-- Executing [s@macro-user-callerid:11] ExecIf("SIP/211-00000014", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/211-00000014", "0?continue") in new stack
-- Executing [s@macro-user-callerid:13] Set("SIP/211-00000014", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/211-00000014", "1?continue") in new stack
-- Goto (macro-user-callerid,s,25)
-- Executing [s@macro-user-callerid:25] Set("SIP/211-00000014", "CALLERID(number)=211") in new stack
-- Executing [s@macro-user-callerid:26] Set("SIP/211-00000014", "CALLERID(name)=330Test") in new stack
-- Executing [*80210@ext-intercom:2] Set("SIP/211-00000014", "dialnumber=210") in new stack
-- Executing [*80210@ext-intercom:3] Set("SIP/211-00000014", "INTERCOM_CALL=TRUE") in new stack
-- Executing [*80210@ext-intercom:4] GotoIf("SIP/211-00000014", "0?end") in new stack
-- Executing [*80210@ext-intercom:5] GotoIf("SIP/211-00000014", "0?end") in new stack
-- Executing [*80210@ext-intercom:6] GotoIf("SIP/211-00000014", "0?allow") in new stack
-- Executing [*80210@ext-intercom:7] GotoIf("SIP/211-00000014", "0?nointercom") in new stack
-- Executing [*80210@ext-intercom:8] GotoIf("SIP/211-00000014", "0?nointercom") in new stack
-- Executing [*80210@ext-intercom:9] Set("SIP/211-00000014", "DEVICES=210") in new stack
-- Executing [*80210@ext-intercom:10] GotoIf("SIP/211-00000014", "0?end") in new stack
-- Executing [*80210@ext-intercom:11] Set("SIP/211-00000014", "LOOPCNT=1") in new stack
-- Executing [*80210@ext-intercom:12] Set("SIP/211-00000014", "_SIPURI=") in new stack
-- Executing [*80210@ext-intercom:13] Set("SIP/211-00000014", "_ALERTINFO=Alert-Info: Ring Answer") in new stack
-- Executing [*80210@ext-intercom:14] Set("SIP/211-00000014", "_CALLINFO=Call-Info: <uri>\;answer-after=0") in new stack
-- Executing [*80210@ext-intercom:15] Set("SIP/211-00000014", "_SIPURI=intercom=true") in new stack
-- Executing [*80210@ext-intercom:16] Set("SIP/211-00000014", "_DOPTIONS=A(beep)") in new stack
-- Executing [*80210@ext-intercom:17] Set("SIP/211-00000014", "_DTIME=5") in new stack
-- Executing [*80210@ext-intercom:18] Set("SIP/211-00000014", "_ANSWERMACRO=") in new stack
-- Executing [*80210@ext-intercom:19] GotoIf("SIP/211-00000014", "0?pagemode") in new stack
-- Executing [*80210@ext-intercom:20] Macro("SIP/211-00000014", "autoanswer,210") in new stack
-- Executing [s@macro-autoanswer:1] Set("SIP/211-00000014", "DIAL=SIP/210") in new stack
-- Executing [s@macro-autoanswer:2] ExecIf("SIP/211-00000014", "0?Set(DIAL=DAHDI/210)") in new stack
-- Executing [s@macro-autoanswer:3] GotoIf("SIP/211-00000014", "0?macro") in new stack
-- Executing [s@macro-autoanswer:4] Set("SIP/211-00000014", "phone=PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134") in new stack
-- Executing [s@macro-autoanswer:5] ExecIf("SIP/211-00000014", "0?Set(CALLINFO=Call-Info: <sip:broadworks.net>\;answer-after=0)") in new stack
-- Executing [s@macro-autoanswer:6] ExecIf("SIP/211-00000014", "1?SipAddHeader(Alert-Info: Ring Answer)") in new stack
-- Executing [s@macro-autoanswer:7] ExecIf("SIP/211-00000014", "1?SipAddHeader(Call-Info: <uri>;answer-after=0)") in new stack
-- Executing [s@macro-autoanswer:8] ExecIf("SIP/211-00000014", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack
-- Executing [*80210@ext-intercom:21] ChanIsAvail("SIP/211-00000014", "SIP/210,s") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*80210@ext-intercom:22] GotoIf("SIP/211-00000014", "0?end") in new stack
-- Executing [*80210@ext-intercom:23] Dial("SIP/211-00000014", "SIP/210,5,A(beep)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 210
-- SIP/210-00000016 is ringing
-- Nobody picked up in 5000 ms
-- Executing [*80210@ext-intercom:24] ExecIf("SIP/211-00000014", "?Return()") in new stack
-- Executing [*80210@ext-intercom:25] Busy("SIP/211-00000014", "20") in new stack
== Spawn extension (ext-intercom, *80210, 25) exited non-zero on 'SIP/211-00000014'
AMG*CLI>Intercom - IP501 -> IP330
AMG*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*80211@from-internal:1] Goto("SIP/210-00000003", "ext-intercom,*80211,1") in new stack
-- Goto (ext-intercom,*80211,1)
-- Executing [*80211@ext-intercom:1] Macro("SIP/210-00000003", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/210-00000003", "AMPUSER=210") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/210-00000003", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/210-00000003", "1?Set(REALCALLERIDNUM=210)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/210-00000003", "AMPUSER=210") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/210-00000003", "AMPUSERCIDNAME=501Test") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/210-00000003", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/210-00000003", "AMPUSERCID=210") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/210-00000003", "CALLERID(all)="501Test" <210>") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/210-00000003", "0?limit") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/210-00000003", "0?Set(GROUP(concurrency_limit)=210)") in new stack
-- Executing [s@macro-user-callerid:11] ExecIf("SIP/210-00000003", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/210-00000003", "0?continue") in new stack
-- Executing [s@macro-user-callerid:13] Set("SIP/210-00000003", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/210-00000003", "1?continue") in new stack
-- Goto (macro-user-callerid,s,25)
-- Executing [s@macro-user-callerid:25] Set("SIP/210-00000003", "CALLERID(number)=210") in new stack
-- Executing [s@macro-user-callerid:26] Set("SIP/210-00000003", "CALLERID(name)=501Test") in new stack
-- Executing [*80211@ext-intercom:2] Set("SIP/210-00000003", "dialnumber=211") in new stack
-- Executing [*80211@ext-intercom:3] Set("SIP/210-00000003", "INTERCOM_CALL=TRUE") in new stack
-- Executing [*80211@ext-intercom:4] GotoIf("SIP/210-00000003", "0?end") in new stack
-- Executing [*80211@ext-intercom:5] GotoIf("SIP/210-00000003", "0?end") in new stack
-- Executing [*80211@ext-intercom:6] GotoIf("SIP/210-00000003", "0?allow") in new stack
-- Executing [*80211@ext-intercom:7] GotoIf("SIP/210-00000003", "0?nointercom") in new stack
-- Executing [*80211@ext-intercom:8] GotoIf("SIP/210-00000003", "0?nointercom") in new stack
-- Executing [*80211@ext-intercom:9] Set("SIP/210-00000003", "DEVICES=211") in new stack
-- Executing [*80211@ext-intercom:10] GotoIf("SIP/210-00000003", "0?end") in new stack
-- Executing [*80211@ext-intercom:11] Set("SIP/210-00000003", "LOOPCNT=1") in new stack
-- Executing [*80211@ext-intercom:12] Set("SIP/210-00000003", "_SIPURI=") in new stack
-- Executing [*80211@ext-intercom:13] Set("SIP/210-00000003", "_ALERTINFO=Alert-Info: Ring Answer") in new stack
-- Executing [*80211@ext-intercom:14] Set("SIP/210-00000003", "_CALLINFO=Call-Info: <uri>\;answer-after=0") in new stack
-- Executing [*80211@ext-intercom:15] Set("SIP/210-00000003", "_SIPURI=intercom=true") in new stack
-- Executing [*80211@ext-intercom:16] Set("SIP/210-00000003", "_DOPTIONS=A(beep)") in new stack
-- Executing [*80211@ext-intercom:17] Set("SIP/210-00000003", "_DTIME=5") in new stack
-- Executing [*80211@ext-intercom:18] Set("SIP/210-00000003", "_ANSWERMACRO=") in new stack
-- Executing [*80211@ext-intercom:19] GotoIf("SIP/210-00000003", "0?pagemode") in new stack
-- Executing [*80211@ext-intercom:20] Macro("SIP/210-00000003", "autoanswer,211") in new stack
-- Executing [s@macro-autoanswer:1] Set("SIP/210-00000003", "DIAL=SIP/211") in new stack
-- Executing [s@macro-autoanswer:2] ExecIf("SIP/210-00000003", "0?Set(DIAL=DAHDI/211)") in new stack
-- Executing [s@macro-autoanswer:3] GotoIf("SIP/210-00000003", "0?macro") in new stack
-- Executing [s@macro-autoanswer:4] Set("SIP/210-00000003", "phone=PolycomSoundPointIP-SPIP_330-UA/3.3.2.0413") in new stack
-- Executing [s@macro-autoanswer:5] ExecIf("SIP/210-00000003", "0?Set(CALLINFO=Call-Info: <sip:broadworks.net>\;answer-after=0)") in new stack
-- Executing [s@macro-autoanswer:6] ExecIf("SIP/210-00000003", "1?SipAddHeader(Alert-Info: Ring Answer)") in new stack
-- Executing [s@macro-autoanswer:7] ExecIf("SIP/210-00000003", "1?SipAddHeader(Call-Info: <uri>;answer-after=0)") in new stack
-- Executing [s@macro-autoanswer:8] ExecIf("SIP/210-00000003", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack
-- Executing [*80211@ext-intercom:21] ChanIsAvail("SIP/210-00000003", "SIP/211,s") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*80211@ext-intercom:22] GotoIf("SIP/210-00000003", "0?end") in new stack
-- Executing [*80211@ext-intercom:23] Dial("SIP/210-00000003", "SIP/211,5,A(beep)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 211
-- SIP/211-00000005 is ringing
-- SIP/211-00000005 answered SIP/210-00000003
-- <SIP/211-00000005> Playing 'beep.gsm' (language 'en')
-- Packet2Packet bridging SIP/210-00000003 and SIP/211-00000005
== Spawn extension (ext-intercom, *80211, 23) exited non-zero on 'SIP/210-00000003'
AMG*CLI>I used ExamDiff and the two are exactly the same line for line sans the extension number. I am hoping someone can make a fool out of me and point out my failure here? 
Thanks to anyone with input. Questions/Comments/Flames accepted graciously.
-Tyler
PS.
Here is the pastebin for my sip_317.cfg if anyone is feeling crazy enough to look at it. http://pastebin.com/8YTjd66e