similar to: Info 16.13.0 with SNOM FW 8.7.5.35

Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit : > Hi, If you're not using RTP encryption did you uncheck the option in your RTP TAB from identity ? > > This is the log. ex dialling 0 from snom phone > > > <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 > <http://123.231.72.210:33878> ---> > INVITE sip:0 at 54.206.59.252

Forgot on the Snom 200 it won't reboot if under the Memory tab in the web interface, Connections > 0 then remote reboot is not possible. Manually cycling the power allows the phone to be rebooted by Sipsak remotely. HOWTO: Reboot a Snom with Sipsak Checklist: 1. Under Advanced in the web interface, is Network Identity set to 5060? 2. Under Advanced in the web interface, is Challenge for

Hello, we just installed the latest 13 and 16 version of asterisk and face problem on incoming calls: they are ended like in Asterisk 16 [2020-02-04 19:19:48] ERROR[3768][C-00000001]: stasis_bridges.c:199 bridge_topics_init: Bridge id initialization required [2020-02-04 19:19:48] WARNING[3768][C-00000001]: bridge.c:809 bridge_base_init: Bridge da3bd3d1-cdea-4a05-8b3d-0ded8c561c5f: Could not

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click "Re-register" in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani

Does someone know coincidentally the cause for the error message specified in the Subject? The following scenario: Snom 360 behind one rout (wiederrum on a DSL line with static IP address hangs). The Snom has a private IP, routs accomplishes NAT. STUN and ICE are activated, as SIP haven 5060/udp are firmly used. Detailed packages passed on on haven 5060/udp of rout to the Snom. The telephone

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

PH> I am setting up a snom 360, and the lights come on OK when the mapped PH> user makes an outgoing call, but when the user takes an incoming call PH> the light does not come on. PH> I do not want to install the bristuff patch if possible. PH> (although I can see that with the devstate command I can make the lights PH> do whatever I want) First, ensure that the 360 has

Hi all, I have been struggling desperately to get * work together with my snom100 for days on end, but I am not making any progress... Of the entries marked *#) I'm still not sure what it does; so far I have on the snom in "SIP/lines" -user name - empty *1) -account - Conrad -registrar - 192.168.200.83 -action - "None" *2) in

Done, ASTERISK-28725 <https://issues.asterisk.org/jira/browse/ASTERISK-28725> but against version 16,7.0 and 13.30.0 as the new ones aren't available in the list Le 04/02/2020 à 19:30, Joshua C. Colp a écrit : > On Tue, Feb 4, 2020 at 2:28 PM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hello, > > we just

Hey all, I've recently updated an Asterisk installation to 11.10.2 (from the Gentoo packaged build), and the update has (seemingly) broken TLS transport for snom phones (I could reproduce on snom 3x0 and 7x0 with 8.4.3x and 8.7.x.y firmwares). What happens is that the registration to the PBX is successful, and inbound calls work normally (i.e., when the phone is being signalled that a

The Asterisk Development Team would like to announce the release of Asterisk 16.13.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.13.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.13.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.13.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

We have a client that has sporadic problems with the *8 pickup facility. The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they get a forbidden message on the phone and I can see the following in the logs. [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL [Aug 8 11:51:53] ERROR[19314] astobj2.c:

I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration

I have been given the following settings for connecting to a voip provider. The names of the fields match my snom phone, and when configured, the phone both makes and recives phonecalls without issue. I am trying to put the same values in asterisk, but there seems to be one field that doesn't seem to exist in asterisk - that of outbound proxy all suggestions welcome SIP headings account

Hi, I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001

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

Anyone have a suggestion for a configuration example using Asterisk and SNOM 190 SIP phone? I have read both sets of documentation and for the life of me, I can't get the phone to register and work. I can use a IAX softphone and it works perfectly. It is just the SIP thing I guess. Anyone have conf examples they can share? Thanks

I recently got a SNOM 360 and have been trying to get the extension lights to work. I can see the subscriptions with sip show subscriptions but I don't see any notifies when a call is made. I must be missing something because I've tried looking to see if anyone else has had this problem but the only solutions I've seen have been to put hints in and I have those. Any suggestions?