similar to: SNOM Phones Displays NR Frequently

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, I've had a Snom 300 connected to my Asterisk box at home for 12 months or so now. Recently it lost all its settings and I had to reconfigure it via the built in website. For a few weeks it was fine. Couple of days ago it lost its settings again. I logged in to its web server and thought I would upgrade the firmware. It seems to be running an old version: Phone Type: snom300-SIP

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

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

-----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

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

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

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 >

Hi All, I realise this is off topic, but its likely the best place to ask! I sent an email to snom support a few days ago but have yet to recieve a response.. Perhaps some one has found a solution to this problem already? I've searched the mailing lists and google and found nothing useful. I've also read Snom's mass deployment documentation but thats no real help in this case.

Under Advanced make sure this is set: Call join on Xfer (2 calls): to off ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tommaso Calosi Sent: Thursday, May 04, 2006 4:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unwanted conference with snom320 and asterisk

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

Hi, Apologies in advance if this is rather long winded ... I have a Windows 2000 DHCP/FTP/TFTP server, serving up a linux-based diags package using PXELinux. Everything is working fine, just like I'd expect. I also have some WinImage images of DOS floppy disks that I'd also like to serve via PXE. So I made up some simple menus using menu.c32 to serve both the floppy image and the

Good day all I have asterisk on a box with one network card I have a 2 companies setup on the system. To keep all apart I binded a different ip to the interface,i,o,w eth0 192.168.0.254 and eth0:1 192.168.1.254 And in sip.conf I took the bind setting out So each company's phones are on a different ip range,and all worked well So we decide to pull the snom190 out and exchange it with a snom360

Hi all, One of my users reports frequently hearing echo on her Snom 360 phone, even while talking to other Snom phones (via Asterisk) on the same LAN (i.e., all-digital low-latency connection). I can never reproduce it though, and swapping the phone didn't help. Has anyone else seen "mystery echo" on Snom phones? Any suggestions for debugging? On my own Snom 360, I sometimes

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

Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on

Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Thanks. Voip-Crazy

Anyone know how I can push a firmware update to a Snom 190 without using DHCP? In the web interface, I specify a path to the Snom firmware, and it works, except I have to physically press OK to get the update to download. I need to do it remotely...

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

I have been doing some reading about gtalk and asterisk. Most of it is pointed to enable using gtalk for making phonecalls. Would it be possible to use gtalk instant messaging input (just some text send to the gtalk account configured on an asterisk box) into the dialplan. This way you could use gtalk im to trigger all kind of events like sending an sms, adding sip entries to the system,