09-13-2013 08:23 AM - edited 03-21-2019 07:46 AM
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
Solved! Go to Solution.
09-17-2013 09:20 AM
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.
09-13-2013 02:00 PM
May be I just don't understand your question correctly, but
09-13-2013 09:11 PM
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
09-13-2013 11:31 PM
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.
09-14-2013 03:04 AM
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>7006>
To: <>>
From: <7006>;tag=3eb19b6c7006>
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=3eb19b6c7006>
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=3eb19b6c7006>
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>7006>
To: <>>
From: <7006>;tag=3eb19b6c7006>
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>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=3eb19b6c7006>
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=3eb19b6c7006>
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=3eb19b6c7006>
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=3eb19b6c7006>
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>7006>
To: <>;tag=as2f227118>
From: <7006>;tag=3eb19b6c7006>
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>7006>
To: <>;tag=as2f227118>
From: <7006>;tag=3eb19b6c7006>
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>7006>
To: <>;tag=as2f227118>
From: <7006>;tag=3eb19b6c7006>
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: --
09-14-2013 06:23 AM
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:
09-17-2013 08:07 AM
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
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=5826b042144b7d5do08001>
To: <>>
Call-ID: aec7f1e6-31a08809@192.168.100.242
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "CISCO" <8001>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=5826b042144b7d5do08001>
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=5826b042144b7d5do08001>
To: <>;tag=as09f233a1>
Call-ID: aec7f1e6-31a08809@192.168.100.242
CSeq: 101 ACK
Max-Forwards: 70
Contact: "CISCO" <8001>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=5826b042144b7d5do08001>
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>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 to8001>
[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=as1ae3104c8001>
To: "CISCO" <8001>;tag=4b051e1ec62e863d8001>
Contact: <8001>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
---
[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>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=5826b042144b7d5do08001>
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=5826b042144b7d5do08001>
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=4b051e1ec62e863d8001>
From: <8001>;tag=as1ae3104c8001>
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=5826b042144b7d5do08001>
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>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: --
[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=5826b042144b7d5do08001>
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=5826b042144b7d5do08001>
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 to8001>
[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=as1ae3104c8001>
To: "CISCO" <8001>;tag=4b051e1ec62e863d8001>
Contact: <8001>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
---
[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=4b051e1ec62e863d8001>
From: <8001>;tag=as1ae3104c8001>
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
09-17-2013 09:20 AM
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.
09-17-2013 01:28 PM
Thank you!!!!
09-17-2013 01:56 PM
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.
09-17-2013 09:43 AM
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.
09-17-2013 01:29 PM
Yes, I will do that. Thank you
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