Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
Community Member

call voicemail extension

Hi,

I use spa502g ip phone with spa500s connect to an asterisk server. I want to know, how I can call the voicemail of extension 7999 directly without the choice of the extension? I try to insert

fnc=sd+cp+blf;sub=*987999@pbxIp

in the attendant console, but it's the same than when I call *98 without the number of extension...

Thank you and sorry for my bad english

1 ACCEPTED SOLUTION

Accepted Solutions
VIP Blue

Re: call voicemail extension

Ok, I try with fnc=sd;ext=*987999@myIp but it still doesn't works, I always be redirect to simple *98 service.

7.5.2

You should consider to upgrade firmware to something more recent.

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Everything is clear now. It seems you should teach something about "Dial Plan" configuration option. If only one pattern match and has been matched completely, the number is considered complete and dialed. Let's allow me to analyze your case number by number:

Digits collected so far
Dial Plan matching state
*Only one pattern matches, but not completely. Wait for more digits (up to Interdigit_Long_Timeout)
*9

Only one pattern matches, but not completely. Wait for more digits (up to Interdigit_Long_Timeout)

*98

Only one pattern matches and matches completely (the *xx). Number completed. Dialing comitted

Rest of number si ignored.

Try this Dial Plan:

(*xx|*98xxxx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Also note this excelent document: Explaining Dial Plans

Mark response as correct answer if it solve your problem.

11 REPLIES
VIP Blue

call voicemail extension

May be I just don't understand your question correctly, but

  1. why are you configuring sub=*987999 when you want to call maibox and mailbox number is 7999 not *987999 ?
  2. why you are not using button dedicated to access to mailbox ? See number 7 on picture below. Note that phone number of mailbox should be configured into "Voice Mail Number" option of "Phone" tab section "General"

http://motioncommunications.co.uk/wp-content/uploads/SPA-502G.jpg

Community Member

Re: call voicemail extension

Hi,
The *98 number is used to enter in the general asterisk voicemail, for example to access user 7999 voicemail I call *987999 that is voicemail number + user number. When I call this number with any internal phone I can access the user 7999 voicemail, but if I register in the attendant console fnc=sd+cp+blf;sub=*987999@pbxIp
I get the same thing than when I call the only *98 number,
so I enter in the general voicemail, not in the user 7999 one.
I already use the Cisco voicemail default button to call the number *97 to access the cisco extension voicemail, user 7999 is for general messages

Thank you

Sent from Cisco Technical Support iPhone App

VIP Blue

Re: call voicemail extension

OK

I assume you deconfigured *98 as service code of blind transfer in the phone (tab "regional" section "Vertical Service Activation Codes")

What I don't know is the detail of implementation within asterisk. Access code *98 7999 may mean either true access number or access code *98 followed by 7999 transmitted as DTMF. If you don't know what I'm speaking about catch the SIP communication between phone and Asterisk during call to general messages voicemail. The INVITE message is the message that interests me.


Community Member

Re: call voicemail extension

This is the log during a call between extension 7006 and 7999 voicemail make from my iphone with zoiper application, thank you for your help

<--- SIP read from UDP:192.168.1.27:45045 --->

INVITE sip:*987999@192.168.1.31;transport=UDP SIP/2.0

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-22c15f9e82446736-1---d8754z-;rport

Max-Forwards: 70

Contact: <7006>

To: <>

From: <7006>;tag=3eb19b6c

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE

Content-Type: application/sdp

Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri

User-Agent: Zoiper r19016

Allow-Events: presence, kpml

Content-Length: 238

v=0

o=Z 0 0 IN IP4 myExternalIp

s=Z

c=IN IP4 myExternalIp

t=0 0

m=audio 32538 RTP/AVP 3 110 98 8 0 101

a=rtpmap:110 speex/8000

a=rtpmap:98 iLBC/8000

a=fmtp:98 mode=20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

<------------->

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: --- (14 headers 12 lines) ---

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: Sending to 192.168.1.27:45045 (NAT)

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Sending to 192.168.1.27:45045 (NAT)

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Using INVITE request as basis request - ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found peer '7006' for '7006' from 192.168.1.27:45045

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c:

<--- Reliably Transmitting (NAT) to 192.168.1.27:45045 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-22c15f9e82446736-1---d8754z-;received=192.168.1.27;rport=45045

From: <7006>;tag=3eb19b6c

To: <>;tag=as2c451c2d

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 1 INVITE

Server: FPBX-2.11.0(11.4.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="552f73f9"

Content-Length: 0

<------------>

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Scheduling destruction of SIP dialog 'ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.' in 6400 ms (Method: INVITE)

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:192.168.1.27:45045 --->

ACK sip:*987999@192.168.1.31;transport=UDP SIP/2.0

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-22c15f9e82446736-1---d8754z-;rport

Max-Forwards: 70

To: <>;tag=as2c451c2d

From: <7006>;tag=3eb19b6c

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 1 ACK

Content-Length: 0

<------------->

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: --- (8 headers 0 lines) ---

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:192.168.1.27:45045 --->

INVITE sip:*987999@192.168.1.31;transport=UDP SIP/2.0

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-8168df8f25abea3a-1---d8754z-;rport

Max-Forwards: 70

Contact: <7006>

To: <>

From: <7006>;tag=3eb19b6c

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 INVITE

Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE

Content-Type: application/sdp

Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri

User-Agent: Zoiper r19016

Authorization: Digest username="7006",realm="asterisk",nonce="552f73f9",uri="sip:*987999@192.168.1.31;transport=UDP",response="614b5226b49bc2f4f39fd5707f6480b9",algorithm=MD5

Allow-Events: presence, kpml

Content-Length: 238

v=0

o=Z 0 0 IN IP4 myExternalIp

s=Z

c=IN IP4 myExternalIp

t=0 0

m=audio 32538 RTP/AVP 3 110 98 8 0 101

a=rtpmap:110 speex/8000

a=rtpmap:98 iLBC/8000

a=fmtp:98 mode=20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

<------------->

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: --- (15 headers 12 lines) ---

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Sending to 192.168.1.27:45045 (NAT)

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Using INVITE request as basis request - ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found peer '7006' for '7006' from 192.168.1.27:45045

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] netsock2.c: == Using SIP RTP TOS bits 184

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] netsock2.c: == Using SIP RTP CoS mark 5

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found RTP audio format 3

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found RTP audio format 110

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found RTP audio format 98

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found RTP audio format 8

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found RTP audio format 0

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found RTP audio format 101

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found audio description format speex for ID 110

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found audio description format iLBC for ID 98

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Found audio description format telephone-event for ID 101

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Peer audio RTP is at port myExternalIp:32538

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: Looking for *987999 in from-internal (domain 192.168.1.31)

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c: list_route: hop: <7006>

[2013-09-14 11:53:34] VERBOSE[1833][C-0000004c] chan_sip.c:

<--- Transmitting (NAT) to 192.168.1.27:45045 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-8168df8f25abea3a-1---d8754z-;received=192.168.1.27;rport=45045

From: <7006>;tag=3eb19b6c

To: <>

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 INVITE

Server: FPBX-2.11.0(11.4.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <>

Content-Length: 0

<------------>

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] pbx.c: -- Executing [*987999@from-internal:1] Answer("SIP/7006-00000080", "") in new stack

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] chan_sip.c: Audio is at 10026

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] chan_sip.c: Adding codec 100003 (ulaw) to SDP

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] chan_sip.c: Adding codec 100004 (alaw) to SDP

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] chan_sip.c: Adding codec 100002 (gsm) to SDP

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] chan_sip.c:

<--- Reliably Transmitting (NAT) to 192.168.1.27:45045 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-8168df8f25abea3a-1---d8754z-;received=192.168.1.27;rport=45045

From: <7006>;tag=3eb19b6c

To: <>;tag=as2f227118

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 INVITE

Server: FPBX-2.11.0(11.4.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <>

Content-Type: application/sdp

Require: timer

Content-Length: 278

v=0

o=root 24814030 24814030 IN IP4 192.168.1.31

s=Asterisk PBX 11.4.0

c=IN IP4 192.168.1.31

t=0 0

m=audio 10026 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

<------------>

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.27:45045:

SIP/2.0 200 OK

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-8168df8f25abea3a-1---d8754z-;received=192.168.1.27;rport=45045

From: <7006>;tag=3eb19b6c

To: <>;tag=as2f227118

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 INVITE

Server: FPBX-2.11.0(11.4.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <>

Content-Type: application/sdp

Require: timer

Content-Length: 278

v=0

o=root 24814030 24814030 IN IP4 192.168.1.31

s=Asterisk PBX 11.4.0

c=IN IP4 192.168.1.31

t=0 0

m=audio 10026 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

---

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:212.27.52.5:5060 --->

Cirpack KeepAlive Packet

<------------->

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.27:45045:

SIP/2.0 200 OK

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-8168df8f25abea3a-1---d8754z-;received=192.168.1.27;rport=45045

From: <7006>;tag=3eb19b6c

To: <>;tag=as2f227118

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 INVITE

Server: FPBX-2.11.0(11.4.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <>

Content-Type: application/sdp

Require: timer

Content-Length: 278

v=0

o=root 24814030 24814030 IN IP4 192.168.1.31

s=Asterisk PBX 11.4.0

c=IN IP4 192.168.1.31

t=0 0

m=audio 10026 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

---

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:192.168.1.27:45045 --->

ACK sip:*987999@192.168.1.31:5060 SIP/2.0

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-262de34169506106-1---d8754z-;rport

Max-Forwards: 70

Contact: <7006>

To: <>;tag=as2f227118

From: <7006>;tag=3eb19b6c

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 ACK

User-Agent: Zoiper r19016

Authorization: Digest username="7006",realm="asterisk",nonce="552f73f9",uri="sip:*987999@192.168.1.31;transport=UDP",response="614b5226b49bc2f4f39fd5707f6480b9",algorithm=MD5

Content-Length: 0

<------------->

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: --- (11 headers 0 lines) ---

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:192.168.1.27:45045 --->

ACK sip:*987999@192.168.1.31:5060 SIP/2.0

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-262de34169506106-1---d8754z-;rport

Max-Forwards: 70

Contact: <7006>

To: <>;tag=as2f227118

From: <7006>;tag=3eb19b6c

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 ACK

User-Agent: Zoiper r19016

Authorization: Digest username="7006",realm="asterisk",nonce="552f73f9",uri="sip:*987999@192.168.1.31;transport=UDP",response="614b5226b49bc2f4f39fd5707f6480b9",algorithm=MD5

Content-Length: 0

<------------->

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: --- (11 headers 0 lines) ---

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:192.168.1.27:45045 --->

ACK sip:*987999@192.168.1.31:5060 SIP/2.0

Via: SIP/2.0/UDP myExternalIp:45045;branch=z9hG4bK-d8754z-262de34169506106-1---d8754z-;rport

Max-Forwards: 70

Contact: <7006>

To: <>;tag=as2f227118

From: <7006>;tag=3eb19b6c

Call-ID: ZjhjZjhhYWVlZjYyODg0NGQ1Y2IzMzgzYjZkNzlhMDk.

CSeq: 2 ACK

User-Agent: Zoiper r19016

Authorization: Digest username="7006",realm="asterisk",nonce="552f73f9",uri="sip:*987999@192.168.1.31;transport=UDP",response="614b5226b49bc2f4f39fd5707f6480b9",algorithm=MD5

Content-Length: 0

<------------->

[2013-09-14 11:53:34] VERBOSE[1833] chan_sip.c: --- (11 headers 0 lines) ---

[2013-09-14 11:53:34] VERBOSE[589][C-0000004c] pbx.c: -- Executing [*987999@from-internal:2] Wait("SIP/7006-00000080", "1") in new stack

[2013-09-14 11:53:35] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:212.27.52.5:5060 --->

Cirpack KeepAlive Packet

<------------->

[2013-09-14 11:53:35] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:212.27.52.5:5060 --->

Cirpack KeepAlive Packet

<------------->

[2013-09-14 11:53:35] VERBOSE[1833] chan_sip.c:

<--- SIP read from UDP:212.27.52.5:5060 --->

Cirpack KeepAlive Packet

<------------->

[2013-09-14 11:53:35] VERBOSE[589][C-0000004c] pbx.c: -- Executing [*987999@from-internal:3] Macro("SIP/7006-00000080", "get-vmcontext,7999") in new stack

[2013-09-14 11:53:35] VERBOSE[589][C-0000004c] pbx.c: -- Executing [s@macro-get-vmcontext:1] Set("SIP/7006-00000080", "VMCONTEXT=default") in new stack

[2013-09-14 11:53:35] VERBOSE[589][C-0000004c] pbx.c: -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/7006-00000080", "0?200:300") in new stack

[2013-09-14 11:53:35] VERBOSE[589][C-0000004c] pbx.c: -- Goto (macro-get-vmcontext,s,300)

[2013-09-14 11:53:35] VERBOSE[589][C-0000004c] pbx.c: -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/7006-00000080", "") in new stack

[2013-09-14 11:53:35] VERBOSE[589][C-0000004c] pbx.c: -- Executing [*987999@from-internal:4] VoiceMailMain("SIP/7006-00000080", "7999@default") in new stack

[2013-09-14 11:53:35] VERBOSE[589][C-0000004c] file.c: -- Playing 'vm-youhave.alaw' (language 'fr')

VIP Blue

Re: call voicemail extension

OK, so *987999 is the complete access number. Now we need to see INVITE packet generated by your phone when you push the button on console. It should generate INVITE to *987999 as well. If not, it's because:

  1. incorrect button definition (try fnc=sd;ext=*987999@pbxIp instead)
  2. *98 is defined as special prefix somewhere or incorrect dial plan - we need to inspect configuration of phone, e.g. content of
    http:///admin/spacfg.xml
Community Member

Re: call voicemail extension

Ok, I try with fnc=sd;ext=*987999@myIp but it still doesn't works, I always be redirect to simple *98 service.

This is my spacfg.xml

Static IP

192.168.100.242

ciscospa502g

255.255.255.0

192.168.100.254

212.27.40.240

212.27.40.241

SPA502G

CCQ17040C92

7.5.2

1.0.4

E02F6D629A01

Installed

Open

None

9/17/2013 15:56:22

00:13:44

4

168

670

67029

0

0

301

48160

211

33760

40

19601

39

22350

N/A

100M Full Duplex

Link Down

Registered

9/17/2013 15:54:21

113 s

No

Idle

None

G711u

G711u

Outbound

No

*98

00:00:01

301

211

48160

33760

70 ms

0 ms

Not Available

0 ms

0

0

0

0

0 ms

0 ms

Idle

None

Idle

Not Installed

Not Installed

Not Installed

Yes

80

Yes

SIP

Yes

No

No

Normal

Static IP

192.168.100.242

255.255.255.0

192.168.100.254

ciscospa502g

212.27.40.240

212.27.40.241

Manual

Parallel

No

0

No

Yes

No

Yes

Yes

3

1

No Limit

No

1

70

5

2

$VERSION

$VERSION

application/dtmf-relay

application/hook-flash

No

No

No

No

No

No

No

No

Yes

No

Yes

5060

5080

No

PAID-RPID-FROM

x-sipura

No

No

No

No

.5

4

5

16

16

16

16

16

240

30

1

7200

30

1200

10

7200

10

10001

10040

0.020

0

0

No

No

No

101

98

97

2

96

99

112

113

G711u

telephone-event

PCMU

PCMA

G726-16

G726-24

G726-32

G726-40

G729a

G729ab

G722

encaprtp

No

No

No

No

No

No

No

No

15

No

No

224.168.168.168:6061

Yes

none

Yes

Yes

Yes

2

600

3600

3600

14400

Yes

Yes

Yes

/spa$PSN.cfg

66,160,159,150,60,43,125

https

$PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH

$PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH

$PN $MAC -- Resync failed: $ERR

Yes

Yes

3600

$PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH

$PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR

$PN $MAC -- Upgrade failed: $ERR

350@-19,440@-19;10(*/0/1+2)

420@-16;10(*/0/1)

520@-19,620@-19;10(*/0/1+2)

480@-19,620@-19;10(.5/.5/1+2)

480@-19,620@-19;10(.25/.25/1+2)

480@-10,620@0;10(.125/.125/1+2)

440@-19,480@-19;*(2/4/1+2)

440@-10;30(.3/9.7/1)

600@-16;1(.25/.25/1)

985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)

914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)

350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)

600@-19;25(.1/.1/1,.1/.1/1,.1/9.5/1)

350@-19;20(.1/.1/1,.1/9.7/1)

397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

600@-16;.3(.05/0.05/1)

600@-19;.2(.05/0.05/1)

440@-10;30(.3/9.7/1)

60(2/4)

60(.3/.2,1/.2,.3/4)

60(.8/.4,.8/4)

60(.4/.2,.3/.2,.8/4)

60(.2/.2,.2/.2,.2/.2,1/4)

60(.2/.4,.2/.4,.2/4)

60(4.5/4)

60(0.25/9.75)

60(.4/.2,.4/2)

255

1800

30

.5

10

3

*69

*66

*86

*72

*73

*90

*91

*92

*93

*56

*57

*71

*70

*67

*68

*81

*82

*77

*87

*78

*79

*16

*17

*18

*19

*96

*38

*36

*39

*37

*03

*017110

*027110

*017111

*027111

*01722

*02722

*0172616

*0272616

*0172624

*0272624

*0172632

*0272632

*0172640

*0272640

*01729

*02729

GMT+01:00

Yes

Yes

-16

.1

12dB

ISO-8859-1

en-US

CISCOSPA502G

CISCOSPA502G

*97

Default

Text Logo

Auto

No

300

Background Picture

1

$USER

private

2

Scrollable

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

No

Yes

1

No

n=Classic-1;w=3;c=1

n=Classic-2;w=3;c=2

n=Classic-3;w=3;c=3

n=Classic-4;w=3;c=4

n=Simple-1;w=2;c=1

n=Simple-2;w=2;c=2

n=Simple-3;w=2;c=3

n=Simple-4;w=2;c=4

n=Simple-5;w=2;c=5

n=Office;w=4;c=1

n=Pulse;w=5;c=1

n=Du-dut;w=6;c=1

0

0

0

0

0

0

pggrp=224.168.168.168:34560;name=All;num=800;listen=yes;

No

Enterprise

No

None

Trusted

No

No

em_login|1;acd_login|1;acd_logout|1;astate|2;avail|3;unavail|3;redial|5;dir|6;cfwd|7;dnd|8;lcr|9;pickup|10;gpickup|11;unpark|12;em_logout

lcr|1;miss|4

redial|1;dir|2;cfwd|3;dnd|4;lcr|5;unpark|6;pickup|7;gpickup|8;starcode|11;alpha|12

dial|1;delchar|2;clear|3;cancel|4;left|5;right|6;starcode|7;alpha|8;dir

endcall|2

hold|1;endcall|2;conf|3;xfer|4;toggle;bxfer;confLx;xferLx;park;phold;flash;

hold|1;endcall|2;xfer|4;toggle;

hold|1;endcall|2;conf|3;toggle;

hold|1;endcall|2;join|4

endcall|2;

resume|1;endcall|2;newcall|3;redial;dir;cfwd;dnd

answer|1;ignore|2;toggle|4

newcall|1;barge|2;cfwd|3;dnd|4

resume|1;barge|2;cfwd|3;dnd|4

Yes

private

3600

No

No

No

$NOTIFY

$PROXY

0x68

3

0xb8

6

high

up and down

UDP

5060

No

Yes

No

4

No

0

none

0

No

No

Yes

Yes

none

No

No

No

No

4

86400

No

No

No

Yes

No

No

192.168.100.240

No

Yes

Yes

No

300

No

No

No

3600

Normal

No

CISCO

8001

*************

No

G711u

No

G711a

Unspecified

Yes

Yes

Yes

Yes

Yes

Yes

Yes

Yes

No

Auto

0

0

No

Default

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Yes

No

20

Yes

No

No

No

No

No

Yes

Speaker

No

12hr

month/day

Yes

Yes

automatic

source

media

Yes

No

No

Yes

Yes

9

8

10

10

Auto

Default

Yes

8

10 s

1800

30

Yes

1

Yes

Asterisk

No

*8

*68

*88

Yes

12

7

fnc=sd+cp+blf;sub=8001@192.168.100.240

fnc=sd+cp+blf;sub=8008@192.168.100.240

fnc=sd;ext=*987999@192.168.100.240

and this is the log of the call in asterisk

<------------->

[2013-09-17 16:42:40] VERBOSE[32523] chan_sip.c: --- (8 headers 0 lines) ---

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:

<--- SIP read from UDP:192.168.100.242:5060 --->

INVITE sip:*98@192.168.100.240 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-834fc996

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 101 INVITE

Max-Forwards: 70

Contact: "CISCO" <8001>

Expires: 240

User-Agent: Cisco/SPA502G-7.5.2

Content-Length: 397

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE

Supported: replaces

Content-Type: application/sdp

v=0

o=- 3640 3640 IN IP4 192.168.100.242

s=-

c=IN IP4 192.168.100.242

t=0 0

m=audio 10035 RTP/AVP 0 8 2 9 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

<------------->

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (14 headers 18 lines) ---

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Using INVITE request as basis request - aec7f1e6-31a08809@192.168.100.242

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found peer '8001' for '8001' from 192.168.100.242:5060

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c:

<--- Reliably Transmitting (NAT) to 192.168.100.242:5060 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-834fc996;received=192.168.100.242;rport=5060

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>;tag=as09f233a1

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 101 INVITE

Server: FPBX-2.11.0(11.5.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2847dbe8"

Content-Length: 0

<------------>

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Scheduling destruction of SIP dialog 'aec7f1e6-31a08809@192.168.100.242' in 6400 ms (Method: INVITE)

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:

<--- SIP read from UDP:192.168.100.242:5060 --->

ACK sip:*98@192.168.100.240 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-834fc996

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>;tag=as09f233a1

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 101 ACK

Max-Forwards: 70

Contact: "CISCO" <8001>

User-Agent: Cisco/SPA502G-7.5.2

Content-Length: 0

<------------->

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (10 headers 0 lines) ---

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:

<--- SIP read from UDP:192.168.100.242:5060 --->

INVITE sip:*98@192.168.100.240 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-6132ab66

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 102 INVITE

Max-Forwards: 70

Authorization: Digest username="8001",realm="asterisk",nonce="2847dbe8",uri="sip:*98@192.168.100.240",algorithm=MD5,response="f4a1ef5ed0d7e5bec2c603a783fe04ff"

Contact: "CISCO" <8001>

Expires: 240

User-Agent: Cisco/SPA502G-7.5.2

Content-Length: 397

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE

Supported: replaces

Content-Type: application/sdp

v=0

o=- 3640 3640 IN IP4 192.168.100.242

s=-

c=IN IP4 192.168.100.242

t=0 0

m=audio 10035 RTP/AVP 0 8 2 9 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

<------------->

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (15 headers 18 lines) ---

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Using INVITE request as basis request - aec7f1e6-31a08809@192.168.100.242

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found peer '8001' for '8001' from 192.168.100.242:5060

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] netsock2.c: == Using SIP RTP TOS bits 184

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] netsock2.c: == Using SIP RTP CoS mark 5

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 0

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 8

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 2

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 9

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 18

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 96

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 97

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 98

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 101

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format PCMU for ID 0

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format PCMA for ID 8

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format G726-32 for ID 2

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format G722 for ID 9

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format G729a for ID 18

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found unknown media description format G726-40 for ID 96

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found unknown media description format G726-24 for ID 97

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found unknown media description format G726-16 for ID 98

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format telephone-event for ID 101

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Peer audio RTP is at port 192.168.100.242:10035

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Looking for *98 in from-internal (domain 192.168.100.240)

[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: set_destination: Parsing <8001> for address/port to send to

[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: set_destination: set destination to 192.168.100.242:5060

[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: Reliably Transmitting (NAT) to 192.168.100.242:5060:

NOTIFY sip:8001@192.168.100.242:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK5b2eaf36;rport

Max-Forwards: 70

From: <8001>;tag=as1ae3104c

To: "CISCO" <8001>;tag=4b051e1ec62e863d

Contact: <8001>

Call-ID: 892e5072-ae14d7f9@192.168.100.242

CSeq: 103 NOTIFY

User-Agent: FPBX-2.11.0(11.5.0)

Subscription-State: active

Event: dialog

Content-Type: application/dialog-info+xml

Content-Length: 207

confirmed

---

[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: == Extension Changed 8001[ext-local] new state InUse for Notify User 8001

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: list_route: hop: <8001>

[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c:

<--- Transmitting (NAT) to 192.168.100.242:5060 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-6132ab66;received=192.168.100.242;rport=5060

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 102 INVITE

Server: FPBX-2.11.0(11.5.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: <>

Content-Length: 0

<------------>

[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:1] Answer("SIP/8001-00000008", "") in new stack

[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Audio is at 10032

[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Adding codec 100003 (ulaw) to SDP

[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Adding codec 100004 (alaw) to SDP

[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP

[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c:

<--- Reliably Transmitting (NAT) to 192.168.100.242:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-6132ab66;received=192.168.100.242;rport=5060

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>;tag=as466725aa

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 102 INVITE

Server: FPBX-2.11.0(11.5.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: <>

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 2104859674 2104859674 IN IP4 192.168.100.240

s=Asterisk PBX 11.5.0

c=IN IP4 192.168.100.240

t=0 0

m=audio 10032 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

<------------>

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:

<--- SIP read from UDP:192.168.100.242:5060 --->

SIP/2.0 200 OK

To: "CISCO" <8001>;tag=4b051e1ec62e863d

From: <8001>;tag=as1ae3104c

Call-ID: 892e5072-ae14d7f9@192.168.100.242

CSeq: 103 NOTIFY

Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK5b2eaf36

Server: Cisco/SPA502G-7.5.2

Content-Length: 0

<------------->

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (8 headers 0 lines) ---

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:

<--- SIP read from UDP:192.168.100.242:5060 --->

ACK sip:*98@192.168.100.240:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-e8d91c8c

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>;tag=as466725aa

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 102 ACK

Max-Forwards: 70

Authorization: Digest username="8001",realm="asterisk",nonce="2847dbe8",uri="sip:*98@192.168.100.240",algorithm=MD5,response="f4a1ef5ed0d7e5bec2c603a783fe04ff"

Contact: "CISCO" <8001>

User-Agent: Cisco/SPA502G-7.5.2

Content-Length: 0

<------------->

[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (11 headers 0 lines) ---

[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:2] Wait("SIP/8001-00000008", "1") in new stack

[2013-09-17 16:42:47] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:3] NoOp("SIP/8001-00000008", "app-dialvm: Asking for mailbox") in new stack

[2013-09-17 16:42:47] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:4] Read("SIP/8001-00000008", "MAILBOX,vm-login,,,3,2") in new stack

[2013-09-17 16:42:47] VERBOSE[32747][C-00000006] file.c: -- Playing 'vm-login.gsm' (language 'fr')

[2013-09-17 16:42:50] VERBOSE[32523] chan_sip.c: Really destroying SIP dialog '7a755620-6e12d2fa@192.168.100.150' Method: REGISTER

[2013-09-17 16:42:51] VERBOSE[32523] chan_sip.c: Really destroying SIP dialog 'dcd8f7d2-aeeb4cf0@192.168.100.150' Method: REGISTER

[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c:

<--- SIP read from UDP:192.168.100.242:5060 --->

BYE sip:*98@192.168.100.240:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-def5f005

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>;tag=as466725aa

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 103 BYE

Max-Forwards: 70

Authorization: Digest username="8001",realm="asterisk",nonce="2847dbe8",uri="sip:*98@192.168.100.240:5060",algorithm=MD5,response="8a4e6470356a8e1ea82eb36413e682cf"

User-Agent: Cisco/SPA502G-7.5.2

Content-Length: 0

<------------->

[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c: --- (10 headers 0 lines) ---

[2013-09-17 16:42:52] VERBOSE[32523][C-00000006] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)

[2013-09-17 16:42:52] VERBOSE[32523][C-00000006] chan_sip.c: Scheduling destruction of SIP dialog 'aec7f1e6-31a08809@192.168.100.242' in 6400 ms (Method: BYE)

[2013-09-17 16:42:52] VERBOSE[32523][C-00000006] chan_sip.c:

<--- Transmitting (NAT) to 192.168.100.242:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-def5f005;received=192.168.100.242;rport=5060

From: "CISCO" <8001>;tag=5826b042144b7d5do0

To: <>;tag=as466725aa

Call-ID: aec7f1e6-31a08809@192.168.100.242

CSeq: 103 BYE

Server: FPBX-2.11.0(11.5.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>

[2013-09-17 16:42:52] VERBOSE[32747][C-00000006] app_read.c: -- User disconnected

[2013-09-17 16:42:52] VERBOSE[32747][C-00000006] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/8001-00000008", "") in new stack

[2013-09-17 16:42:52] VERBOSE[32747][C-00000006] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/8001-00000008'

[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: set_destination: Parsing <8001> for address/port to send to

[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: set_destination: set destination to 192.168.100.242:5060

[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: Reliably Transmitting (NAT) to 192.168.100.242:5060:

NOTIFY sip:8001@192.168.100.242:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK60445d1d;rport

Max-Forwards: 70

From: <8001>;tag=as1ae3104c

To: "CISCO" <8001>;tag=4b051e1ec62e863d

Contact: <8001>

Call-ID: 892e5072-ae14d7f9@192.168.100.242

CSeq: 104 NOTIFY

User-Agent: FPBX-2.11.0(11.5.0)

Subscription-State: active

Event: dialog

Content-Type: application/dialog-info+xml

Content-Length: 208

terminated

---

[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: == Extension Changed 8001[ext-local] new state Idle for Notify User 8001

[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c:

<--- SIP read from UDP:192.168.100.242:5060 --->

SIP/2.0 200 OK

To: "CISCO" <8001>;tag=4b051e1ec62e863d

From: <8001>;tag=as1ae3104c

Call-ID: 892e5072-ae14d7f9@192.168.100.242

CSeq: 104 NOTIFY

Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK60445d1d

Server: Cisco/SPA502G-7.5.2

Content-Length: 0

<------------->

[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c: --- (8 headers 0 lines) ---

[2013-09-17 16:42:58] VERBOSE[32523] chan_sip.c: Really destroying SIP dialog 'aec7f1e6-31a08809@192.168.100.242' Method: BYE

[2013-09-17 16:43:02] VERBOSE[32744] asterisk.c: -- Remote UNIX connection disconnected

Thank you

VIP Blue

Re: call voicemail extension

Ok, I try with fnc=sd;ext=*987999@myIp but it still doesn't works, I always be redirect to simple *98 service.

7.5.2

You should consider to upgrade firmware to something more recent.

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Everything is clear now. It seems you should teach something about "Dial Plan" configuration option. If only one pattern match and has been matched completely, the number is considered complete and dialed. Let's allow me to analyze your case number by number:

Digits collected so far
Dial Plan matching state
*Only one pattern matches, but not completely. Wait for more digits (up to Interdigit_Long_Timeout)
*9

Only one pattern matches, but not completely. Wait for more digits (up to Interdigit_Long_Timeout)

*98

Only one pattern matches and matches completely (the *xx). Number completed. Dialing comitted

Rest of number si ignored.

Try this Dial Plan:

(*xx|*98xxxx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Also note this excelent document: Explaining Dial Plans

Mark response as correct answer if it solve your problem.

Community Member

Re: call voicemail extension

Thank you!!!!

VIP Blue

Re: call voicemail extension

Nice to hear it work now.

Note that you should develop your own Dial Plan that fit your local numbering plan as well as outside national calls and international calls. The default Dial Plan is designed for US and may not fit your needs.

VIP Blue

Re: call voicemail extension

By the way, the standard codec for the France PSTN is aLaw (PCMA), so you should prefer it against uLaw. uLaw calls terminated in PSTN need transcoding which may distort a sound somewhat.

You should change the configuration of your phones as well as the configuration of FreePBX.

Community Member

Re: call voicemail extension

Yes, I will do that. Thank you

763
Views
0
Helpful
11
Replies
CreatePlease to create content