search for: snom190

...fering party's caller ID on a BT101? Otherwise I will have to trash the 101's (I hate these garbage phones anyway, but it would be nice for customers if they're able to choose between 101's and more expensive 190's). Using asterisk 1.0.7 + bristuff RC8g; version 3.56m on the snom190's and version 1.0.6.3 on the BT101's. regards, Elwin Andriol

Hi, I have searched and searched, but cannot identify what is happening here... I have several snom190 phones, and all of them have the 5th function key set to call "asterisk" by using the "destination" option. This automatically causes the phone to SUBSCRIBE for NOTIFY messages for the "asterisk" destimation. Asterisk does not seem to like the SUBSCRIBE requests. It l...

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GrandStream102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels they end up being transferred to...

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels and the first caller hangs up wh...

.... They only think they are." ----- Original Message ----- From: "Matt Kemner" <zombie@penguincare.com.au> To: "David Wilson" <dave@dcdata.co.za> Sent: Thursday, March 10, 2005 12:57 PM Subject: Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190 > On Thu, 10 Mar 2005, quoth David Wilson: > >> Sorry about the misleading subject :) I started a couple days ago being >> very >> unclear about how things were going wrong and thought it could be >> something >> in Asterisk that was causing it. > > Yea...

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...

...feature working on some snom 190 phones but having no luck... As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. I've email'd snom a few days ago but have yet to get a response. On my 190s, im running snom190-SIP 3.57v. I am pulling the config for the phone from a web page. Im setting: intercom_enabled&: on Looking in the settings.htm page on the phone verifies this, as does the listing in the advanced.htm page. Any one have any ideas why I'm having this problem? Cheers, -Shaun <>==...

I have a number of internal SIP phones and a number of external SIP clients with the server running Asterisk on the boundary between the two. ie the server has two network cards with an internal private address and an external public address. For security reasons no routing is allowed between the two thus no internal phone can talk directly to external phone or visa versa. I am very happy with

...rd 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 This company is on the virtual interface(eth0:1) The 360 register and can make outgoing calls fine But when you try to make a call to it it does not work? I gives this error in the cli "Forbidden - wrong password on authentication for INVITE to '"301...

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 CVS-01/29/04-16:41:27 I would appreciate any help to fix the NR problem. Thanks

...om: "snom_01" <sip:snom_01@192.168.0.129>;tag=i7u8p4i1vi To: "snom_01" <sip:snom_01@192.168.0.129> Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 CSeq: 45683 REGISTER Max-Forwards: 70 Contact: <sip:snom_01@192.168.102.70:5060;line=v8ppcao5>;q=1.0 User-Agent: snom190-3.56i P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.102.70 WWW-Contact: <http://192.168.102.70:80> WWW-Contact: <https://192.168.102.70:443> Expires: 60 Content-Length: 0 ------------------------------------------------------------------------ Received from...

Greetings All- I have an * box with the NuFone H.323 channel driver installed. I also have an Altigen VoIP system with a PRI to the PSTN. I can sucessfully make a call from a SIP extension (snom190) to an H.323 extension (altigen phone) The thing I can't seem to make work is a call from a H.323 phone to a SIP extension. Here's the layout: <snom190{1235}>==[192.168.0.0/24]==<asterisk>==[IPSEC VPN 192.168.0.0/24 - 10.0.0.0/24] ==<altigen server>==[10.0.0.0/24]==<...

This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of them didn't work well. The symptom is that outgoing RTP audio is garbled - like the packets are pulsed. Inbound is fin...

...0.129>;tag=i7u8p4i1vi >>To: "snom_01" <sip:snom_01@192.168.0.129> >>Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 >>CSeq: 45683 REGISTER >>Max-Forwards: 70 >>Contact: <sip:snom_01@192.168.102.70:5060;line=v8ppcao5>;q=1.0 >>User-Agent: snom190-3.56i >>P-NAT-Refresh: 15 >>Supported: gruu >>Allow-Events: dialog >>X-Real-IP: 192.168.102.70 >>WWW-Contact: <http://192.168.102.70:80> >>WWW-Contact: <https://192.168.102.70:443> >>Expires: 60 >>Content-Length: 0 >> >>-------...

Here is a part of a working sip.conf for Asterisk with SNOM190 Phones. Try using host=dynamic and have a closer look at the configuration in the snom 190. Also, try using dtmfmode=rfc2833 . [general] realm = hallinux2.gwsnettech.local port = 5060 bindaddr = 0.0.0.0 context = default disallow=all allow=alaw allow=ulaw allow=gsm register => 081503:xxxxxx@sip...

...TW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I update a snom phone?" The phone used is a snom 190 (snom190-SIP 3.52e). If I use the webinterface to insert the URL it works fine, but I am not able to set this URL using dhcp. -- Stefan Tichy <asterisk@pi4tel.de>

Hi, I am using a number of snom190 phones, and an asterisk "gateway" server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended...

I am just playing with a SNOM 190. Overall, I'm very impressed with the quality of the unit and the feature set. I am running the latest firmware (snom190-SIP 3.57u) and the asterisk CVS from last night (1/3/05). The only problem that I've encountered so far is with Call Forwarding, which doesn't work at all. The Snom phone is sending a "486 - Busy Here" back to *, which results in the call going to voicemail (as per the dialplan)...

...as the one channel model seems to work well (see http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk). The setup is such: --------- Inet --> VoIP provider ---> POTS | | (iax2, NAT) | asterisk (on abox with iptables fw) | (SIP, LAN) |----------> SNOM190 phones | ----------> SIP-GSM-module ---> SIM cards --> mobile phone networks Sound, however, is too bad for the SIP to GSM module to be usable. Call initiation from the LAN to the GSM network works but the audio stream stops and continues for about 1 to 3 seconds each, irregularl...

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 th...