* You are viewing the archive for August 15th, 2011

Asterisk -> Office 365 Unified Messaging… anyone done it?

Trying to make this work, and Office 365 support is useless, giving me the following response when I asked them for help troubleshooting a 488 Not Acceptable Here.

Regarding your service request about configuring your PBX system with Office 365, we do not support specific setups for PBX systems for Unified Messaging. Please contact the vendor for more specific instructions and configurations.

Here is a SIP debug:

[2011-08-11 23:00:26] VERBOSE[17000] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061:
OPTIONS sip:um.outlook.com SIP/2.0
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162
Max-Forwards: 70
From: "Unknown" ;tag=as438c582c
To:
Contact:
Call-ID: <a href="mailto:67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4">67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4</a>:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.5.0)
Date: Fri, 12 Aug 2011 06:00:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0 —
[2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c:
< — SIP read from TLS:65.55.174.100:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162
From: "Unknown" ;tag=as438c582c
To: ;tag=b4ec76231
Call-ID: <a href="mailto:67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4">67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4</a>:5061
CSeq: 102 OPTIONS
ACCEPT: application/sdp
CONTENT-LENGTH: 0
ALLOW: INVITE
ALLOW: BYE
ALLOW: CANCEL
ALLOW: OPTIONS
ALLOW: ACK
ALLOW: INFO
ALLOW: NOTIFY
SERVER: RTCC/3.5.0.0 < ————->
[2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c: — (16 headers 0 lines) —
[2011-08-11 23:00:27] VERBOSE[17000] chan_sip.c: Really destroying SIP dialog ’67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4:5061′ Method: OPTIONS
[2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Audio is at 5061
[2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061:
INVITE sip:999@um.outlook.com SIP/2.0
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02
Max-Forwards: 70
From: "Test User" ;tag=as746bc17a
To:
Contact:
Call-ID: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>:5061
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.5.0)
Date: Fri, 12 Aug 2011 06:00:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 238 v=0
o=root 1381221379 1381221379 IN IP4 1.2.3.4
s=Asterisk PBX 1.8.5.0
c=IN IP4 1.2.3.4
t=0 0
m=audio 17688 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv —
[2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c:
< — SIP read from TLS:65.55.174.100:5061 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02
From: "Test User" ;tag=as746bc17a
To:
Call-ID: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>:5061
CSeq: 102 INVITE
Content-Length: 0 < ————->
[2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: — (7 headers 0 lines) —
[2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c:
< — SIP read from TLS:65.55.174.100:5061 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02
From: "Test User" ;tag=as746bc17a
To: ;tag=aprqngfrt-hm4td720000c6
Call-ID: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>:5061
CSeq: 102 INVITE
Content-Length: 0 < ————->
[2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: — (7 headers 0 lines) —
[2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: Transmitting (no NAT) to 65.55.174.100:5061:
ACK sip:999@um.outlook.com SIP/2.0
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02
Max-Forwards: 70
From: "Test User" ;tag=as746bc17a
To: ;tag=aprqngfrt-hm4td720000c6
Contact:
Call-ID: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>:5061
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.5.0)
Content-Length: 0 —
[2011-08-11 23:00:48] VERBOSE[17000] chan_sip.c: Really destroying SIP dialog ‘535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061′ Method: INVITE

This could be an unsupported codec. Digging through some Cisco documentation linked to as a guide for configuring CCM 8.0 with Office 365, it states that they support 711ulaw . I also tried setting dtmfmode=auto/rfc2833/info/inband with no luck.

Trying to get someone with a brain at MS to work with me on this.

TIA

SIP trunk trouble. Please help.

Hello,

I am troubleshooting a SIP trunk problem. The system is Asterisk 1.8.5.
The problem is can’t make any outbound/inbound. It always get “Number is
not valid 701″.

I tried to figure out the reason the call got dropped and couldn’t find
out the solution. I noticed that in the SIP debug there are two IP (from
the provider) involved:
209.205.85.162
209.205.85.130

It seems my Asterisk sent INVITE to the first IP, but the provider want
use 2nd one. How can I make it works? I never seen this thing before.
(BTW, if I test this account on a Linsys ATA it works just fine!)

Here is my sip.conf setting and the debug out put.

Thanks for help!

Jian

Linksys/Cisco 504G randomly restarts

I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
to be random. Sometimes as short as 6 minutes.
FW version is 7.4.3a

I have searched and tried disabling FW check and all related settings.
I also extended all the default 3600 resync checks to a lot longer.

TIA
CF

issue with some numbers

Hello List,

i have a small issue regarding to some numbers when i call these numbers
using asterisk i got all times answer_machine ,but when i call these
numbers from my phone or any numbers i get the customer and i can speak
without any issue. The issue just when i call these numbers using my
system asterisk.

Any help will be appreciated

Thanks and regards

app_swift for Asterisk 10

Hey there folks,

I’d sent this to the list last night and got reject email this
morning. Apparently it is always a good idea to have an active
subscription to the list you are trying to post to – just one of those
things. :)

In any case, a new beta version of app_swift is available for Asterisk
10. I put it up in the Asterisk Forge on the 25th of last month, but
wanted to wait to post something on the users list until I’d had a
chance to really test it a bit (so far so good).

http://forge.asterisk.org/gf/project/app_swift/frs/

I have to say, the combination of Asterisk 10 and this latest version
of app_swift is absolutely the best sounding of any release to-date!
I’ve been *very* impressed so far.

Also, just fyi . . there are some extremely minor tweaks I’ll be
back-porting to the other app_swift versions shortly. I hope to get
that done this weekend or next depending on my free time.

Enjoy,

– Darren

asterisk-users Digest, Vol 85, Issue 23

Numtodial is the variable that will receive the DTMF input – enternum would
be the prompt played before entry (/var/lib/asterisk/sounds/enternum.wav
(gsm, slin, whatever the codec dictates).

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tahar .H
Sent: Saturday, August 13, 2011 7:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 23

hi folks,

can some one please explain to me what this one stands for :

Exten => 1234,1,read(numtodial,enternum,10,skip,1,10)

that numtodial and enternum !!!!