Attendent Directed Call Pickup "Unknown Party"

[Anonymous]

I've searched and searched but have not found a solution to my problem.

Using Asterisk 1.6.2.16.1
Polycom 3.3.1.0933

The BLF shows up fine but when I try to do a pickup by the "Pickup" softkey or by pressing the extension directly, it displays "Unknown Party" and not pickup the line.

The only thing I see missing that was in many suggestions is that "notifycid=yes" is not in place. CFCUTILITY always strips this out. And inputting it manually does not change results.

Any ideas?

sip.cfg

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<polycomConfig xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="polycomConfig.xsd">
<attendant>
<attendant.behaviors>
<attendant.behaviors.display>
<attendant.behaviors.display.spontaneousCallAppear ances attendant.behaviors.display.spontaneousCallAppeara nces.automata="1" />
</attendant.behaviors.display>
</attendant.behaviors>
<attendant.resourceList attendant.resourceList.1.type="automata" />
</attendant>
<bg>
<bg.medRes>
<bg.medRes.gray>
<bg.medRes.gray.bm bg.medRes.gray.bm.1.name="custom.jpg" />
</bg.medRes.gray>
</bg.medRes>
</bg>
<call call.directedCallPickupMethod="legacy" call.directedCallPickupString="**" />
<dialplan dialplan.applyToCallListDial="0" dialplan.digitmap="*xxxxx|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" />
<divert divert.1.contact="01234" />
<feature>
<feature.directedCallPickup feature.directedCallPickup.enabled="1" />
<feature.enhancedFeatureKeys feature.enhancedFeatureKeys.enabled="1" />
<feature.presence feature.presence.enabled="1" />
</feature>
<msg>
<msg.mwi msg.mwi.1.callBack="*97" msg.mwi.1.callBackMode="contact" />
</msg>
<reg>
<reg.1.server reg.1.server.1.address="192.168.168.30" />
</reg>
<voIpProt>
<voIpProt.SIP>
<voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.value="auto-answer" />
<voIpProt.SIP.specialEvent>
<voIpProt.SIP.specialEvent.checkSync voIpProt.SIP.specialEvent.checkSync.alwaysReboot=" 1" />
</voIpProt.SIP.specialEvent>
</voIpProt.SIP>
<voIpProt.server voIpProt.server.1.address="192.168.168.30" />
</voIpProt>
</polycomConfig>

mac.cfg

<?xml version="1.0" standalone="yes"?>
<PHONE_CONFIG>

<OVERRIDES
reg.1.displayName="160=Rick"
reg.1.address="160"
reg.1.label="160-Rick"
reg.1.auth.userId="160"
reg.1.auth.password="xxxxx"
reg.1.type="shared"
/>

<attendant
attendant.resourceList.1.address="sip:117@192.168. 168.30"
attendant.resourceList.1.label="117-Nick"
attendant.resourceList.2.address="sip:140@192.168. 168.30"
attendant.resourceList.2.label="140-Caitlin"
/>

</PHONE_CONFIG>

[SteffenBaierUK]

Hello K9Sports,

welcome to the Polycom Community.

Asterisk related issues should be posted in the Digium Asterisk Forum => here <=

The Parameter notifycid=yes is actually a Asterisk Parameter and is therefore removed by the Polycom cfcUtility.

The feature.presence.enabled is for the presence feature including management of buddies and own status or Microsoft Live Communications Server 2005 and not necessary for the Attendant functionality.

Are you using additional Softkeys or EFK as you have the  feature.enhancedFeatureKeys.enabled="1" activated?

Can you confirm that the string set in call.directedCallPickupString="**" matched the features.conf configuration in Asterisk?

Is the message Unknown Party shown on the Phone Display or in the Asterisk CLI?

The correct way to go forward would be to check the Asterisk CLI in debug mode or get a wireshark trace in order to check what the Polycom Phone actually sends.

In some circumstances the Phone may send the **+Extension and Asterisk just expects to receive the ** alone. 

You could remove this additional Numbers via the Asterisk extension.conf.

We are currently updating the Polycom Asterisk Guide => here <= so future Versions may include more details on this.

Some Parameters I used in my Lab Setup sip.conf in the [general] Section are:

subscribecontext=local

allowsubscribe=yes

notifyringing=yes

notifyhold=yes

limitonpeers=yes

Please be aware that above is only a guideline and may depend on your setup.

In addition the separate [peers] or [general] section could have:

Callgroup=1

pickupgroup=1

and you may want to allow

calls=2

Please see above only as an example and other members of the community may have more details on the setup.

Best Regards

Steffen Baier

Polycom Global Services

[Anonymous]

• Will be using softkeys for various functions but will remove that feature if this is blocking the pickup.
• I confirmed and actually changed it back to *8 in both areas.
• UNKNOWN PARTY is displayed on the phone very briefly.
• Tried removing the additional number, no luck.
• Tried all I could from the cuurent guide
• Added those parameters to sip.conf.  Still no luck.

Anything else I can try?  I've been trying everything people suggests yet still haven't hit the problem.  I know this works because others have it working with the same equipment.  There has to be one little setting I just don't have set correctly.

[SteffenBaierUK]

Hello K9sports,

did you check the Asterisk CLI in a debug verbosity or a Wireshark Trace as recommended?

Best regards

Steffen Baier

Polycom Global Services

[Anonymous]

Sorry... forgot to add that part.  I went thru the logs and didn't even see where the Polycom phone even tries to send a pickup command string.

[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:3] Set("SIP/EE_In-00000021", "__EXTTOCALL=117") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:4] Set("SIP/EE_In-00000021", "__PICKUPMARK=117") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:5] Macro("SIP/EE_In-00000021", "blkvm-setifempty,") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/EE_In-00000021", "1?init") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (macro-blkvm-setifempty,s,4)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-blkvm-setifempty:4] Set("SIP/EE_In-00000021", "__BLKVM_CHANNEL=SIP/EE_In-00000021") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-blkvm-setifempty:5] Set("SIP/EE_In-00000021", "SHARED(BLKVM,SIP/EE_In-00000021)=TRUE") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-blkvm-setifempty:6] Set("SIP/EE_In-00000021", "GOSUB_RETVAL=TRUE") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/EE_In-00000021", "") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:6] GotoIf("SIP/EE_In-00000021", "1?skipov") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (from-did-direct,117,9)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:9] Set("SIP/EE_In-00000021", "RRNODEST=") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:10] Set("SIP/EE_In-00000021", "__NODEST=117") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:11] GosubIf("SIP/EE_In-00000021", "0?sub-fmsetcid,s,1") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:12] Set("SIP/EE_In-00000021", "RecordMethod=Group") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:13] Macro("SIP/EE_In-00000021", "record-enable,117-117,Group") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/EE_In-00000021", "1?check") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (macro-record-enable,s,4)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/EE_In-00000021", "0?MacroExit()") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/EE_In-00000021", "1?Group:OUT") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (macro-record-enable,s,6)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:6] Set("SIP/EE_In-00000021", "LOOPCNT=2") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:7] Set("SIP/EE_In-00000021", "ITER=1") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("SIP/EE_In-00000021", "1?continue") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (macro-record-enable,s,12)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:12] Set("SIP/EE_In-00000021", "ITER=2") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:13] GotoIf("SIP/EE_In-00000021", "1?begin") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (macro-record-enable,s,8)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:8] GotoIf("SIP/EE_In-00000021", "1?continue") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (macro-record-enable,s,12)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:12] Set("SIP/EE_In-00000021", "ITER=3") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:13] GotoIf("SIP/EE_In-00000021", "0?begin") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:14] GotoIf("SIP/EE_In-00000021", "0?IN") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-record-enable:15] ExecIf("SIP/EE_In-00000021", "1?MacroExit()") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:14] Set("SIP/EE_In-00000021", "RingGroupMethod=ringallv2") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:15] Set("SIP/EE_In-00000021", "_FMGRP=117") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:16] GotoIf("SIP/EE_In-00000021", "0?doconfirm") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [117@from-did-direct:17] Macro("SIP/EE_In-00000021", "dial,20,tr,117") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-dial:1] GotoIf("SIP/EE_In-00000021", "1?dial") in new stack
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Goto (macro-dial,s,3)
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-dial:3] AGI("SIP/EE_In-00000021", "dialparties.agi") in new stack
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: dialparties.agi: Caller ID name is '1111111111' number is '1111111111'
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: dialparties.agi: Methodology of ring is 'ringallv2'
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: -- dialparties.agi: Added extension 117 to extension map
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: -- dialparties.agi: Extension 117 cf is disabled
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: -- dialparties.agi: Extension 117 do not disturb is disabled
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: -- dialparties.agi: dbset CALLTRACE/117 to 1111111111
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: -- dialparties.agi: Filtered ARG3: 117
[Sep 29 20:46:38] VERBOSE[7292] res_agi.c: -- <SIP/EE_In-00000021>AGI Script dialparties.agi completed, returning 0
[Sep 29 20:46:38] VERBOSE[7292] pbx.c: -- Executing [s@macro-dial:7] Dial("SIP/EE_In-00000021", "SIP/117,22,trM(auto-blkvm)") in new stack
[Sep 29 20:46:38] VERBOSE[7292] netsock2.c: == Using SIP RTP TOS bits 184
[Sep 29 20:46:38] VERBOSE[7292] netsock2.c: == Using SIP RTP CoS mark 5
[Sep 29 20:46:38] VERBOSE[3191] chan_sip.c: == Extension Changed 117[ext-local] new state Ringing for Notify User 160
[Sep 29 20:46:38] VERBOSE[3191] chan_sip.c: == Extension Changed 117[ext-local] new state Ringing for Notify User 140
[Sep 29 20:46:38] VERBOSE[7292] app_dial.c: -- Called SIP/117
[Sep 29 20:46:38] VERBOSE[7292] app_dial.c: -- SIP/117-00000022 is ringing
[Sep 29 20:46:43] VERBOSE[3211] netsock2.c: == Using SIP RTP TOS bits 184
[Sep 29 20:46:43] VERBOSE[3211] netsock2.c: == Using SIP RTP CoS mark 5
[Sep 29 20:46:45] VERBOSE[3211] netsock2.c: == Using SIP RTP TOS bits 184
[Sep 29 20:46:45] VERBOSE[3211] netsock2.c: == Using SIP RTP CoS mark 5
[Sep 29 20:46:45] VERBOSE[3191] chan_sip.c: == Extension Changed 160[ext-local] new state InUse for Notify User 140
[Sep 29 20:46:45] VERBOSE[3191] chan_sip.c: == Extension Changed 160[ext-local] new state InUse for Notify User 117
[Sep 29 20:46:45] VERBOSE[3191] chan_sip.c: == Extension Changed 160[ext-local] new state Idle for Notify User 140 (queued)
[Sep 29 20:46:45] VERBOSE[3191] chan_sip.c: == Extension Changed 160[ext-local] new state Idle for Notify User 117 (queued)
[Sep 29 20:46:45] VERBOSE[3211] chan_sip.c: == Extension Changed 160[from-internal] new state Idle for Notify User 140
[Sep 29 20:46:45] VERBOSE[3211] chan_sip.c: == Extension Changed 160[from-internal] new state Idle for Notify User 117
[Sep 29 20:46:47] VERBOSE[7292] app_macro.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/EE_In-00000021' in macro 'dial'
[Sep 29 20:46:47] VERBOSE[7292] pbx.c: == Spawn extension (from-did-direct, 117, 17) exited non-zero on 'SIP/EE_In-00000021'
[Sep 29 20:46:47] VERBOSE[7292] pbx.c: -- Executing [h@from-did-direct:1] Macro("SIP/EE_In-00000021", "hangupcall,") in new stack
[Sep 29 20:46:47] VERBOSE[7292] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/EE_In-00000021", "1?theend") in new stack
[Sep 29 20:46:47] VERBOSE[7292] pbx.c: -- Goto (macro-hangupcall,s,3)
[Sep 29 20:46:47] VERBOSE[7292] pbx.c: -- Executing [s@macro-hangupcall:3] Hangup("SIP/EE_In-00000021", "") in new stack
[Sep 29 20:46:47] VERBOSE[7292] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/EE_In-00000021' in macro 'hangupcall'
[Sep 29 20:46:47] VERBOSE[7292] pbx.c: == Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/EE_In-00000021'
[Sep 29 20:46:47] VERBOSE[3191] chan_sip.c: == Extension Changed 117[ext-local] new state Idle for Notify User 160
[Sep 29 20:46:47] VERBOSE[3191] chan_sip.c: == Extension Changed 117[ext-local] new state Idle for Notify User 140 

[SteffenBaierUK]

Hello K9Sports,

this is really an Asterisk Issue as I have this working as desribed in my detailed reply but I see below as an example for other Asterisk Users in the community.

Checking in the Asterisk CLI via sip set debug <extension/peer> I see:

<--- Transmitting (NAT) to 10.252.75.173:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.252.75.173;branch=z9hG4bK2646da1d1F38CE4A;received=10.252.75.173
From: "710" <sip:710@10.252.75.119>;tag=488143F2-572F70EB
To: <sip:*97@10.252.75.119;user=phone>
Call-ID: df02fa3e-5a71b187-f52ea04c@10.252.75.173
CSeq: 2 INVITE
Server: Steffens Asterisk 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:*97@10.252.75.119>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'df02fa3e-5a71b187-f52ea04c@10.252.75.173' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 10.252.75.173:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.252.75.173;branch=z9hG4bK2646da1d1F38CE4A;received=10.252.75.173
From: "710" <sip:710@10.252.75.119>;tag=488143F2-572F70EB
To: <sip:*97@10.252.75.119;user=phone>;tag=as3469d983
Call-ID: df02fa3e-5a71b187-f52ea04c@10.252.75.173
CSeq: 2 INVITE
Server: Steffens Asterisk 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

Above uses the *97 Pickup Code with the legacy method.

Asterisk is rejecting the call as the single peer had not been added to a pickupgroup and callgroup in the sip.conf [peer] Asterisk section.

Please see below just as an example from my Lab Config and address any further issues directly with the Digium Forum or use your reseller and/or Polycom Support via PPI (Pay Per Incident).

Please be aware that the context etc. matches my local seup so please do not just copy and paste!

sip.conf

[general]
language=en
context=default
context=local
bindport=5060
bindaddr=0.0.0.0
udpbindaddr=0.0.0.0            ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) added for LYNC
tcpenable=yes                  ; Enable server for incoming TCP connections (default is no) added for LYNC
tcpbindaddr=0.0.0.0:5060           ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) added for LYNC
useragent=Steffens Asterisk 1.6.2.18
videosupport=yes
srvlookup=yes
vmexten=mailbox
disallow=all
allow=ulaw
allow=alaw
allow=g722
allow=h263p
allow=h261
allow=h264
dtmfmode=auto
maxexpirey=145
subscribecontext=local
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
;useclientcode=yes
;canreinvite=yes
limitonpeers=yes
progressinband=never
dtmfmode=rfc2833 ; we assume clients are behind NAT

and an example peer within the sip.conf

[700]; Extension 700
domain=0.0.0.0
user=700
type=friend
secret=700
mailbox=700
nat=yes; we assume clients are behind NAT
host=dynamic; and have dynamic IP addresses
callerid="User 700" <700> Name being Displayed on the Far End
allowsubscribe=yes
call-limit=10
context=internal
Callgroup=1
pickupgroup=1

My BLF Configuration

<call call.directedCallPickupMethod="legacy" call.directedCallPickupString="*97" />
<attendant attendant.resourceList.1.address="sip:700@10.252.75.119" attendant.resourceList.1.label="700" attendant.resourceList.2.address="sip:752@10.252.75.119" attendant.resourceList.2.label="752" />

My User Details

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Application SIP Wolverine 3.3.2.0413 11-Aug-11 12:17 -->
<!-- Created 19-09-2011 17:56 -->
<CONFIG>
	<OVERRIDES 
	reg.1.address="710"
		reg.1.auth.password="710"
		reg.1.auth.userId="710"
		reg.1.displayName="710"
		reg.1.label="710"
		reg.1.server.1.address="10.252.75.119"
		up.screenCapture.enabled="1"
				/>
</CONFIG>

Above basic setup allows me to Pickup Calls via the BLF Button when Flashing or calling the Extension when just pressing.

It also shows the current status of the Extension.

Best Regards

Steffen Baier

Polycom Global Services