Dan Lawrence
Member
- Joined
- Jan 4, 2008
- Messages
- 47
- Reaction score
- 9
I'm trying to get paging/auto answer/intercom working and am running into dead ends. I've searched the forums and see other people trying to do the same thing, but none of the suggestions seem to apply to my situation.
I'm running a hosted PIAF version 3.0.6.5 at RentPBX (which I am very happy with).
From everything I can tell, the target phone is getting the right invite commands, but is simply not auto answering the call.
I set up the phones using OSS PBX End Point Manager version 2.11.7 which worked very well. I see the proper config lines " <alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1.class="4"/>" and "<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1" />" are present in the sip_327.cfg file in the /tftpboot folder, so I am as confident as I can be that the phone is seeing the proper settings for auto answer.
I've looked at the syslog output from the phone but it is useless. I get messages about the boot process and what config files it is loading, but nothing about actual call process.
The phones are on firmware version 3.2.7 which Polycom says is the latest version available for the IP430 (http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html)
Polycom release notes for 3.2.6 indicate "68184 Phone no longer sends double confirmation on auto answer." which implies that auto answer is supported.(http://downloads.polycom.com/voice/voip/relnotes/SIP_Software_Release_Notes_3_2_7.pdf)
I don't know what else to check and I'm open to suggestions. Has anyone seen a IP430 actually perform an auto answer?
Version info, TCPDUMP and asterisk debug output is below. Thanks for any ideas you might have.
I'm running a hosted PIAF version 3.0.6.5 at RentPBX (which I am very happy with).
From everything I can tell, the target phone is getting the right invite commands, but is simply not auto answering the call.
I set up the phones using OSS PBX End Point Manager version 2.11.7 which worked very well. I see the proper config lines " <alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1.class="4"/>" and "<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1" />" are present in the sip_327.cfg file in the /tftpboot folder, so I am as confident as I can be that the phone is seeing the proper settings for auto answer.
I've looked at the syslog output from the phone but it is useless. I get messages about the boot process and what config files it is loading, but nothing about actual call process.
The phones are on firmware version 3.2.7 which Polycom says is the latest version available for the IP430 (http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html)
Polycom release notes for 3.2.6 indicate "68184 Phone no longer sends double confirmation on auto answer." which implies that auto answer is supported.(http://downloads.polycom.com/voice/voip/relnotes/SIP_Software_Release_Notes_3_2_7.pdf)
I don't know what else to check and I'm open to suggestions. Has anyone seen a IP430 actually perform an auto answer?
Version info, TCPDUMP and asterisk debug output is below. Thanks for any ideas you might have.
Code:
┌────────────────────────SYSTEM INFORMATION───────────────────────────┐
│ 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 = 3.0.6.5 under *XEN* on Rent PBX │
│ FreePBX Version = 2.11.0.38 │
│ Running Asterisk Version = 11.7.0 │
│ Asterisk Source Version = 11.7.0 │
│ Dahdi Source Version = 2.8.0.1 │
│ Libpri Source Version = 1.4.14 │
│ IP Address = ww.xx.yy.zz on eth0 │
│ Operating System = Scientific Linux release 6.5 │
│ Kernel Version = 2.6.32-431.5.1.el6.i686 - 32 Bit │
│ │
└─────────────────────────────────────────────────────────────────────┘
Code:
PIAFSERVER.IP is the IP address of the PIAF server.
The target phone is extension 501, is at 10.11.12.201 on the local network and is named "Comp".
The calling phone is extension 502 is at 10.11.12.202 on the local network and is named "Bedrm".
The number I dialed was "*80501".
I acually had the route the call through my DISA so I could get the traffic to route though the home router where I could run tcpdump on it.
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK6e504648
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as6b8f3de0
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.7.0)
Date: Sun, 14 Sep 2014 08:23:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK6e504648
From: "Unknown" <sip:[email protected]>;tag=as6b8f3de0
To: "Comp" <sip:[email protected]:5060>;tag=343ADB2E-F3745C25
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK01a97353
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as718d8fd2
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.7.0)
Date: Sun, 14 Sep 2014 08:24:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK01a97353
From: "Unknown" <sip:[email protected]>;tag=as718d8fd2
To: "Comp" <sip:[email protected]:5060>;tag=E2D4D567-CBE1F852
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0
INVITE sip:[email protected]:5060;intercom=true SIP/2.0
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK5342d0d4
Max-Forwards: 70
From: "Bedrm" <sip:[email protected]>;tag=as7f5b3cb4
To: <sip:[email protected]:5060;intercom=true>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.7.0)
Date: Sun, 14 Sep 2014 08:25:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Call-Info: <uri>;answer-after=0
Alert-Info: info=Auto Answer
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 445089591 445089591 IN IP4 PIAFSERVER.IP
s=Asterisk PBX 11.7.0
c=IN IP4 PIAFSERVER.IP
t=0 0
m=audio 10078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK5342d0d4
From: "Bedrm" <sip:[email protected]>;tag=as7f5b3cb4
To: "Comp" <sip:[email protected]:5060;intercom=true>;tag=8FF8B1EC-C66396BB
CSeq: 102 INVITE
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
Accept-Language: en
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK5342d0d4
From: "Bedrm" <sip:[email protected]>;tag=as7f5b3cb4
To: "Comp" <sip:[email protected]:5060;intercom=true>;tag=8FF8B1EC-C66396BB
CSeq: 102 INVITE
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
Call-Info: <uri>;answer-after=0
Allow-Events: talk,hold,conference
Accept-Language: en
Content-Length: 0
CANCEL sip:[email protected]:5060;intercom=true SIP/2.0
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK5342d0d4
Max-Forwards: 70
From: "Bedrm" <sip:[email protected]>;tag=as7f5b3cb4
To: <sip:[email protected]:5060;intercom=true>
Call-ID: [email protected]:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.11.0(11.7.0)
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK5342d0d4
From: "Bedrm" <sip:[email protected]>;tag=as7f5b3cb4
To: "Comp" <sip:[email protected]:5060;intercom=true>
CSeq: 102 CANCEL
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
Accept-Language: en
Content-Length: 0
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK5342d0d4
From: "Bedrm" <sip:[email protected]>;tag=as7f5b3cb4
To: "Comp" <sip:[email protected]:5060;intercom=true>;tag=8FF8B1EC-C66396BB
CSeq: 102 INVITE
Call-ID: [email protected]:5060
Contact: <sip:[email protected]:5060>
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
Accept-Language: en
Content-Length: 0
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP PIAFSERVER.IP:5060;branch=z9hG4bK5342d0d4
Max-Forwards: 70
From: "Bedrm" <sip:[email protected]>;tag=as7f5b3cb4
To: <sip:[email protected]:5060;intercom=true>;tag=8FF8B1EC-C66396BB
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.7.0)
Content-Length: 0