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 chan_sip.c: Reliably Transmitting (no NAT) to 184.108.40.206:5061: OPTIONS sip:um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 220.127.116.11:5061;branch=z9hG4bK16884162 Max-Forwards: 70 From: "Unknown" ;tag=as438c582c To: Contact: Call-ID: email@example.com:5061 CSeq: 102 OPTIONS User-Agent: FPBX-2.8.1(18.104.22.168) Date: Fri, 12 Aug 2011…Asterisk Users 4.2 years ago 0 Answers
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:
22.214.171.124 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.
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
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
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…
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: firstname.lastname@example.org
[mailto:email@example.com] On Behalf Of Tahar .H
Sent: Saturday, August 13, 2011 7:13 PM
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 !!!!
Configure un numero GSM qui est operationel dans ton gateway, les addresse ip
dans chaque tel et numero sip.
Ton gateway au niveau des tel ce l' adresse ip de ton ASTERISK. Neto Dalima Arcene
De : A J Stiles
À : Asterisk Users Mailing List - Non-Commercial Discussion
Envoyé le : Lun 15 août 2011, 10h 09min 29s
Objet : Re: [asterisk-users] DID to display the calling number On Sunday 14 Aug 2011, bilal ghayyad wrote:
> Hi All;