VoIP Mailing List Archives

[nick at flhsi.com]

Greetings all, First time poster, Sorry if this has been answered here before.

We recently replaced a failed 1.4x asterisk PBX at a customer location.


Voicemail access was setup when the customer dialed *8, This worked in 1.4.


Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon).


The strange part is *8 no longer works.
The only CLI feedback I get is "== Using SIP RTP CoS mark 5"


In features.conf, Callpickup *8 is commented out, But just incase I also changed it to *7 (We don't use that feature).


It appears to be something completely SIP based, As if the call originates from DAHDI, It works fine..


If anyone has any ideas, Please let me know. Thanks!


SIP Trace Below


<--- SIP read from UDP:208.65.55.170:5063 --->
INVITE sip:*8@10.65.6.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576
From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868
To: <sip:*8@10.65.6.10>
Call-ID: 695101044@172.16.10.101
CSeq: 1 INVITE
Contact: <sip:nicktest@172.16.10.101:5063>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T46G 28.71.0.180
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 308


v=0
o=- 20402 20402 IN IP4 172.16.10.101
s=SDP data
c=IN IP4 172.16.10.101
t=0 0
m=audio 11792 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv


<------------->
--- (14 headers 15 lines) ---
== Using SIP RTP CoS mark 5
Using INVITE request as basis request - 695101044@172.16.10.101
Found peer 'nicktest' for 'nicktest' from 208.65.55.170:5063
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.10.101:11792
Looking for *8 in trunk_office (domain 10.65.6.10)
list_route: hop: <sip:nicktest@172.16.10.101:5063>


<--- Transmitting (NAT) to 208.65.55.170:5063 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576;received=208.65.55.170
From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868
To: <sip:*8@10.65.6.10>
Call-ID: 695101044@172.16.10.101
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:*8@10.65.6.10>
Content-Length: 0




<------------>
Scheduling destruction of SIP dialog '695101044@172.16.10.101' in 6400 ms (Method: INVITE)


<--- Reliably Transmitting (NAT) to 208.65.55.170:5063 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576;received=208.65.55.170
From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868
To: <sip:*8@10.65.6.10>;tag=as65ceb9be
Call-ID: 695101044@172.16.10.101
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0




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


<--- SIP read from UDP:208.65.55.170:5063 --->
ACK sip:*8@10.65.6.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576
From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868
To: <sip:*8@10.65.6.10>;tag=as65ceb9be
Call-ID: 695101044@172.16.10.101
CSeq: 1 ACK
Content-Length: 0




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

Nick Olsen
Network Operations (855) FLSPEED x106

[vlad at mikhelson.com]

Nick,

You may want to try *97 and *98 to access voice mail.

Regards,
Vladimir


On 12/31/2013 10:23 AM, Nick Olsen wrote:

[wealwildwon at wombit.com]

On 12/31/2013 12:41 PM, Vladimir Mikhelson wrote:

[nick at flhsi.com]

Dialplan is solid..

exten => *8,1,VoicemailMain(@default)
exten => *8,2,Hangup
exten => 88,1,VoicemailMain(@default)
exten => 88,2,Hangup


Also tried "_*8" in the dialplan at the request of a fellow BOFH.


Using 88 temporarily, Which works fine. Also, DAHDI dumps into the same context and has no issue. I did indeed restart the service after any features change. I always run my CLI with about 8 million v's, But still don't get any useful feedback on this issue.


I understand I can easily change the voicemail number. But this customer (hotel) has the voicemail number printed on in-room cards. So I'm hoping not to cause them a re-print.

Nick Olsen
Network Operations (855) FLSPEED x106





From: "Adrian Serafini" <wealwildwon@wombit.com>
Sent: Tuesday, December 31, 2013 12:51 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] *8 and SIP

On 12/31/2013 12:41 PM, Vladimir Mikhelson wrote:

[andres at telesip.net]

On 12/31/13, 11:23 AM, Nick Olsen wrote:

[nick at flhsi.com]

That did it.

For some reason, Even commented out. Pick up was still *8. And persisted even after an asterisk service restart. Changed the feature to *7, Rebooted the whole PBX and it finally took effect.

Nick Olsen
Network Operations (855) FLSPEED x106





From: "Andres" <andres@telesip.net>
Sent: Tuesday, December 31, 2013 2:22 PM
To: nick@flhsi.com, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] *8 and SIP

On 12/31/13, 11:23 AM, Nick Olsen wrote: