similar to: OT: Snom 300 losing config?

Hi, I am using SNOM phones with Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months. Here are SNOM Phone and the firmware version; snom190-SIP - Version-Code: snom190-SIP 3.56m snom320-SIP - snom320 jffs2 v3.36 snom300-SIP - snom300-SIP 6.5.2 Asterisk version - Asterisk

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ====ok inbound and outbound the calls between x-lite and snom300====> ok inbound and

please no body has som with aastra can help me in this issue 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>: > hello list > > i need your help please regarding an issue with snom300 and aastra6731i > using asterisk > > 11.13.0 asterisk > > snom 300 8.7.3.25 > > astra 6731i 2.6.0.2019 > > i have configured the trunks like

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack --

Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841 ~15 Grandstream 101 Everything (currently) is on the same network, not a router to be seen between any two. Also everything, except the snom, is working sweetly. The main problem is ECHO.. awful echo and

Hello All! I am in the process of assembling an asterisk-based phone system for my office - using SPA-3000s to connect the network to the PSTN. I am wondering if anybody else can get (or has already seen) the same behaviour out of their 3000. The short version: Send multiple Calls to the SPA's FXO port at the same time it is re-registering with Asterisk. SPA HTTP Configuration:

Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and

I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command "ping -f -c 10000 192.168.2.10". Outgoing

Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from

On Fri, Feb 19, 2016 at 11:12:32AM +0000, Andrei Perietanu wrote: >I'm trying to clone a volume in a storage pool and I'm following the steps >described here: >http://libvirt.org/docs/libvirt-appdev-guide-python/en-US/html/libvirt_application_development_guide_using_python-Storage_Pools-Cloning.html > >My code looks like: > > destXML = """ >

Can anyone help with BLF for Snom 300s ? (Asterisk 1.4.10.1) I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing

I'm trying to clone a volume in a storage pool and I'm following the steps described here: http://libvirt.org/docs/libvirt-appdev-guide-python/en-US/html/libvirt_application_development_guide_using_python-Storage_Pools-Cloning.html My code looks like: destXML = """ <volume>

On Fri, Feb 19, 2016 at 12:33:10PM +0000, Andrei Perietanu wrote: >On Fri, Feb 19, 2016 at 12:12 PM, Martin Kletzander <mkletzan@redhat.com> >wrote: > >> On Fri, Feb 19, 2016 at 11:12:32AM +0000, Andrei Perietanu wrote: >> >>> I'm trying to clone a volume in a storage pool and I'm following the steps >>> described here: >>> >>>

Hi All, I'm running trixbox 2.0. The problem: a remote extension behind a NAT, can call other extensions, can call any other party, can call voicemail, will ring when rung, but when answered there is nothing and the dialling party continues to hear the ring tone. I'm getting this error in the logs: "That's odd... Got a response on a call we dont know about" I see

Hi, We face a strange behavior with Asterisk 1.8.15 and SIP defaultuser definition. in sip.conf [blabla0](natted-phone,ulaw-phone,callgroup1,snom-320) defaultuser=tel-221 mailbox=221 callerid="My CID" dtmfmode=auto ;defaultip=10.0.12.21 CLI sip show peer blabla0 Addr->IP : 10.0.12.21:2067 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP

On Fri, Feb 19, 2016 at 12:12 PM, Martin Kletzander <mkletzan@redhat.com> wrote: > On Fri, Feb 19, 2016 at 11:12:32AM +0000, Andrei Perietanu wrote: > >> I'm trying to clone a volume in a storage pool and I'm following the steps >> described here: >> >>

Hi there, Is there a utility to create an EXT3 file system image from directories? Just like the mkfs.jffs2 which creates a JFFS2 file system image from directories? The "mke2fs -j" only creates the bare bone file system, what I want is to build an image with pre-built content. Thanks, Debbie