similar to: Why would asterisk presume a loop (482 "Loop Detected")?

I have a issue trying to understand why Asterisk-PBX, when a SNOM (320 or 360) successfully redirects/diverts a call when it is a local extension, but fails when you enter external number. Both the local extension dial and external extension dial are within the same context [from-sip] and both phones are capable of making external calls. I have looked at the standard sites, but not

Hello Although not stictly a asterisk issue, any help would be apreciated. Firstly a few notes on the snom 360, which I have had on a test bed for the last week. Its a great phone, with a good user interface, both physically and its web based one. At its lastest firmware it does have a few quirks, with regards to the way it handles usernames and passwords on the physical interface. These have

Hi, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the following. Any thoughts on whet to check next? Thanks, Steve ### Call comes in here and is answered

10.0.10.136 136 3c267009caa 131 InUse dialog-info+xml Then, I "grep 3c267009caa /var/log/asterisk/full": Feb 20 02:12:09 DEBUG[3180] chan_sip.c: Stopping retransmission on '3c267009caa3-frjgby2n2g25 at snom360-000413239566' of Request 1109: Match Found Feb 20 02:25:28 DEBUG[3180] chan_sip.c: Stopping retransmission on

Here is my setup: * is on a NAT'd subnet, but also has an externally routable IP address. I have a Snom360 that's external to this and behind NAT. The Snom360 can call other phones in * subnet (by their internal extension numbers) and voice is transmitted fine; however, when I attempt to check voicemail (or any * voice recordings for that matter) I can't hear them. The phone just

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

Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3 My SNOM sends the dtmf-relay and Asterisk gets it because I can see it in the sip debug. However, once seen, Asterisk doesn't actually do anything about it. I've checked features and that seems fine. Is this a bug or something that I've screwed up? For the record, here's the features setting: asterisk*CLI>

Sorry if this is an obvious question, but I haven't seen an obvious answer on the wiki that I remember. Has anyone managed to make the record button on the snom 360 fire off the Monitor() application? I don't see a bounty, and googling for "snom 360 record button asterisk" returns tons of product specification pages. (Joy!) I don't see a bounty for it, and the only

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

We are looking to replace our existing Legacy PBX with Asterisk. Our receptionist currently has a light display for a certain extension when someone is on a call. When she needs to transfer she simply hits that button. Is it possible to use a snom360 + Sidecar to monitor 30 extensions and make transfers using the buttons? Does the Cisco 7960 expansion module work with asterisk? Thanks.

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN Cards, if i press in a call the "*" Asterisk, Asterisk destroys the call not, Asterisk lets him hang and do nothing, if i hangup, Asterisk tell me in the warnings-log that the bridging was not successfull ?! If have disabled the function to hangup in the features.conf, but the key is still available, can someone

Hi, I want to pick up a call with the snom's programmable buttons(snom190 -SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried with the option 'Destination' and when the incoming call arrive to another snom phone the button blinking. In this way I can only pick down it pressing the blinking button. The solution is call the *8 or parcking the call but my pbroblem

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

We have a large install of 360's running rev 4.1 with zero problems. I did another, smaller install couple weeks ago with 40 360's running rev 5.3. In both cases, the install was identical, same Asterisk version, same dialplan, everything the same except the differences were: 1. Different firmware rev 2. Different physical LAN Guess what? On the smaller install, lockups and reboots.

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

Hi, I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and, while hearing the dial tone of ringing Phone C, places the handset on hook. Phone B sends a REFER,

Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the