Old/FAQ
From Snom User Wiki
Basics | VoIP
Q: Do I need a PC to connect to the Internet or can I connect the phone to the Internet directly via LAN or dial-up modem?
A: You do not necessarily need a PC but you do require a working internet connection (also) used by the phone. Usually you need a common router that connects Ethernet devices like our phones to the Internet. The phone can be used in conjunction with a PC dialing up to the Internet, but does not have an integrated modem.
Q: What can I do when the IP address assigned by my ISP changes dynamically but infrequently?
A: To keep the phone working, you should register it at a proxy/registrar on the Internet that also remembers the phone’s IP address. When the IP address of the phone changes, re-register the new IP address at that proxy. The expiry date until next re- registering can be set up via web interface-> line X-> SIP:
Proposed Expiry: [1 day | 8 hours | 2 hours | 1 hour | 10 minutes | 1 minute]
There are also several vendors that offer solutions for operating in an environment with frequently changing IP addresses.
Q: Can I use snom phones to make long distance calls via the Internet in order to bypass conventional telephony services?
A: Yes, you can call anyone over IP (e.g. via public Internet) if the called party owns a VoIP phone and is registered at an SIP registrar, e.g. at snom.info. (This is a public server running an SIP registrar.) If the called party does not own a VoIP phone you can still call them via a PSTN gateway that re-routes your call into the PSTN (Plain Old Telephone System). If your provider does not offer this service you will have to set up your own gateway.
Q: How can I update the date/time using a date/time server?
A: On the "Advanced Page" you have to specify a date/time server (e.g. 192.53.103.103) and timezone. The phone obviously needs to have IP connectivity to the time server and it will then fetch the time at the beginning, using the NTP protocol.
Features | Call Establishment | Call Diversion | Phone Book
Q: Do snom phones have Outlook integration?
A: No, you need a TSP (TAPI service provider) that controls a SIP phone in order to integrate Outlook. snom offers an Outlook Add-In which allows dialing phone numbers directly from Outlook Address Book. The Add-In converts the Outlook number format including whitespace, brackets and hyphen into SIP URIs.
Q: Do snom phones provide the "Hotline Functionality"?
A: Yes, the snom190 and snom3xx series support this feature. "Hotline functionality" is meant to work like this: lifting up the handset (hook off) dials a preassigned number e.g. in lobbies etc. The mentioned phones can be configured using Action URL´s on the Action URL page:
On offhook: http://127.0.0.1/command.htm?number=Extension
Extension has to be replaced by the specific extension.
Q: Can I establish pure IP calls between snom Phones?
A: Yes. Please, configure the following settings on each phone first:
Advanced page:
o Filter Packets from Registrar "off" o Listen on port 5060 for TCP "on" o Network Identity: 5060
Rebooting every device should enable the phones for pure IP calls. It was reported that calling from some 3rd party softphones might require the following:
sip:@<IP address>
Note: Your phone does not provide any dialtone before call establishment.
Q: What does call diversion mean?
A: It is a redirection feature. The phone can divert immediately, on busy, after n seconds or never. It is also called "Call Forwarding" or "Call Diversion".
Q: How can I quickly switch call diversion on or off?
A: If the redirection number does not change very frequently do this:
1) Administrator-> Webinterface-> Preferences: Choose the Event and the proper number
2) Administrator-> Webinterface-> Function Keys: Assign the type "Key Event" to any of the available keys (D-keys upper/ P-keys lower will work), enter the string "F_REDIRECT" into the number field
3) Phone User-> Phone-> Pressing the "F_REDIRECT" key will toggle the redirection feature: "CFwd" is displayed when redirection is switched on.
Q: How can I configure call diversion on the snom3xx phone?
A: Follow these steps:
1) Press "Menu" Button
2) With the arrow keys navigate until "Redirection" is selected
3) Press "Enter" Symbol for entering the submenu and once again for changing the redirection mode
4) Use the arrow keys to define whether you want your incoming calls redirected "always", "when busy", "on timeout" or "never"
5) If you decide e.g. for "always" go back into the "redirection" submenu and select "redirect target"
6) This has only be done once or when you want to change the number: Press "enter" and insert a phone extension for the redirection
7) Leave the menu or just wait some seconds until the idle screen re-appaers on the display, "CFwd" is now displayed there.
8) For disabling redirection go to step 4) and select "Never"
Q: Is "auto answering mode" some kind of answering machine?
A: No. It means that the phone picks up the call immediately. This can be useful in call center environments or public address systems.
Q: Is it possible to filter out calls but not using "anonymous call blocking"?
A: Yes, we have a “deny” list where you can enter undesired caller numbers. On the "Advanced" page, section "Phone behaviour" configure the setting
Deny all feature: ON
This will display an additional icon on each incoming call (DND + !). Pressing the softkey below that icon blocks the incoming call and adds the caller to the Deny list (-> Address Book) automatically.
Q: Do snom phones offer integrated auto-attendant/voice response or the ability to download/insert a WAV file?
A: Voice mail is not integrated into the phones. This is a function offered by a Media server/SIP PBX. The phone does alert-info and it can play WAV files as custom ringer melody.
Registrar Interoperability | Asterisk | OpenSER
Q: How to make Asterisk send INVITEs to trigger the phone for Intercom
A: Add something like the following to your extensions.conf:
exten => 452,1,SIPAddHeader(Call-Info:<sip:domain>\;answer-after=0)
"domain" has to be replaced by the domain used by you.
Q: Calling the mailbox from the phone doesn't work. How to solve this?
A: Have a look at the SIP NOTIFY Asterisk is sending to the phone, delivering the voicemail status. In that NOTIFY message Asterisk is adding a SIP URI which is used by the phone for retrieving the message (calling the mbox). Most probably in your case the SIP URI is not correct. Do you have already set up the following in your Asterisk?
sip.conf: mailbox=<your Mailboxnumber> extensions.conf: exten => asterisk,1,VoiceMailMain()
The NOTIFY sent from * informs the phone about how many messages are waiting and the voice mail extension to call:
NOTIFY ... CSeq: 1 NOTIFY ... Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: yes Message-Account: sip:mailboxextension@your_asterisk_domain Voice-Message: 2/0 (0/0)
The phone acknowledges this message:
SIP/2.0 200 Ok ... CSeq: 1 NOTIFY Content-Length: 0
Q: Why is hotdesking not working with Asterisk?
A: The problem seems to be that Asterisk is not sending 401/407, but 403 after the authentication failed, 403 clearly means: Stop talking to me! There seems to be a patch available.
Network | SIP | RTP/SRTP | IP/TCP/UDP | Routing/NAT | Codec
Q: Why my snom phone needs minutes before it gets registered after boot up?
A: Customers with such problems reported they were having this problem especially with managable switches from e.g. HP, Cisco, Linksys, etc, which are trying to negotiate with the phones regarding STP, CDP, 802.1x, etc. Switching this off resolved the issue for them.
In case of using STP (Spanning Tree Protocol) RSTP (Rapid Spanning Tree Protocol) may be used, which decreases the connection pause below a second. If RSTP isn't available, STP should be switched off for that port.
Q: How can I enable media encryption on my snom phone?
A: To enable encryption on your snom phone please set up the following parameter:
Line page RTP tab:
RTP Encryption to "on"
Turning this setting to "on" will enable the snom phone to offer/answer media encryption parameters. It is important to note that the phone will only resort to secure media (SRTP) if the peer UA also supports the same set of media encryption protocols. The presence of a "Lock" symbol on the snom phone display indicates an encrypted call.
Please note: snom190 and snom3XX use different incompatible encryption methods:
snom190: RFC3261, k header snom360: RFC3711 SRTP ecncryption algorithm AES
AES is implemeted as described in this document:
http://www.cs.columbia.edu/sip/drafts/mmusic/draft-ietf-mmusic-sdescriptions-09.txt
Q: What is the required IP bandwidth for each SIP call session?
A: See this article for a complete description:
Q: Can I obtain statistical information on packets from the phone during a call?
A: Yes, it is possible via RTCP. If you turn on RTCP support in the advanced webpage of the phone, the phone will send standard RTCP Sender Reports to the other phone periodically. They contain the relevant call information including packet loss, fraction lost, jitter, timestamps info. and other information.
Links:
Q: What does the TOS value 160 mean?
A: TOS significates "Type of service" and represents the 2nd byte in the IP datagramm. TOS usually refers to "Quality of Service" in packet networks.
Traditionally, the first three IP precedence (-> RFC 791) bits were supposed to be used in TOS Application Routing (RFC 1583-> OSPF, IS-IS) but no application really supports it. The TOS field has then been redefined as the Differentiated Services Code Point (DSCP-> RFC 2474) which consists of the first 6 bits and 2 bits used for a TCP mechanism called Explicit Congestion Notification (ECN) defined in RFC 3168.
The value 160 (Binary 10100000) means IP precedence 5 (Binary 101) or DSCP Class Selector 5 (Binary 101000 = 0x28). In order to obtain low-delay, low-jitter, low-loss service Expedited Forwarding should be used instead (= 184):
| IP Prec | IP Prec Bin | DSCP Class | DSCP Bin | DSCP Hex | DCSP Dec | TOS value (snom) |
|---|---|---|---|---|---|---|
| 0 | 000 | Best Effort | 000000 | 0x00 | 0 | 0 |
| 1 | 001 | CS 1 | 001000 | 0x08 | 8 | 32 |
| AF11-Low | 001010 | 0x0A | 10 | 40 | ||
| AF12-Medium | 001100 | 0x0C | 12 | 48 | ||
| AF13-High | 001110 | 0x0E | 14 | 56 | ||
| 2 | 010 | CS 2 | 010000 | 0x10 | 16 | 64 |
| AF21-Low | 010010 | 0x12 | 18 | 72 | ||
| AF22-Medium | 010100 | 0x14 | 20 | 80 | ||
| AF23-High | 010110 | 0x16 | 22 | 88 | ||
| 3 | 011 | CS 3 | 011000 | 0x18 | 24 | 96 |
| AF31-Low | 011010 | 0x1A | 26 | 104 | ||
| AF32-Medium | 011100 | 0x1C | 28 | 112 | ||
| AF33-High | 011110 | 0x1E | 30 | 120 | ||
| 4 | 100 | CS 4 | 100000 | 0x20 | 32 | 128 |
| AF41-Low | 100010 | 0x22 | 34 | 136 | ||
| AF42-Medium | 100100 | 0x24 | 36 | 144 | ||
| AF43-High | 100110 | 0x26 | 38 | 152 | ||
| 5 | 101 | CS 5 | 101000 | 0x28 | 40 | 160 |
| Expedited Fwdg | 101110 | 0x2E | 46 | 184 |
Links:
Q: How can the phone look up potential numbers to dial on a remote server?
A: This can be done via SIP PUBLISH messages. When the digits/characters to dial are being typed in, the phone sends a PUBLISH request to the proxy. Assume we have typed in 1 and 0:
PUBLISH sip:101@192.168.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.198:2051;branch=z9hG4bK-gp8d9ozwdmzx;rport From: <sip:101@192.168.0.1:5060>;tag=d60pqeilvc To: <sip:101@192.168.0.1:5060> Call-ID: 3c278cdab239-330c9hntldee@snom320-000413240068 CSeq: 1 PUBLISH Max-Forwards: 70 Event: number-guessing Content-Type: application/text Content-Length: 25 Number: 10 Max-Hits: 3
The proxy sends back three alternatives attached to the answer reply:
SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.6.198:2051;branch=z9hG4bK-gp8d9ozwdmzx;rport=2051 f: <sip:101@192.168.0.1:5060>;tag=d60pqeilvc t: <sip:101@192.168.0.1:5060> i: 3c278cdab239-330c9hntldee@snom320-000413240068 CSeq: 1 PUBLISH c: application/number-guessing l: 136 "Franky Chang" <sip:101@192.168.0.1;user=phone> "Steven Jones" <sip:102@192.168.0.1;user=phone> "Marie Sun" <sip:103@192.168.0.1;user=phone>
Q: How to change the displayed source and destination data of a call?
A: This can be done via the following SIP INFO message. The message must be sent during the existing dialog.
INFO sip:123@192.168.4.185:2051;line=z6yxt6av SIP/2.0 v: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-c5036dd36491b35dba0953bcd912b2a7;rport f: <sip:*66@intern.snom.de;user=phone>;tag=24433 t: <sip:456@intern.snom.de;transport=udp>;tag=ga2hjuhxot i: 3c268afd0ea6-tk5ueq9zigfb@snom360-000413230077 CSeq: 1 INFO Max-Forwards: 70 m: <sip:*66@192.168.0.8:5060;transport=udp> c: message/sipfrag l: 68 From: <sip:456@intern.snom.de> To: "123" <sip:123@intern.snom.de>
Q: How to show billing information on the phone display?
A: Please read "Advice of charge (AOC) in SIP"
Q: What does "Broken Registrar" mean?
A: When incoming INVITEs from your provider do not contain the Contact URI which was previously registered by your phone as its "Contact", the phone cannot safely identify the target line of the incoming call. When you compare the URI in the first line of the incoming INVITE and the URI in the Contact of the REGISTER the phone sends to the registrar of your provider, they will probably differ. In this case, the registrar is said to be "broken".
Non-technically spoken, if your provider works only when you turn on "Support broken registrar" it means your provider does not call your phone the way the phone has requested to be called. It is as though your provider adressed a letter to an apartment building with the city, street, and house number, but without the apartment number on it. If you turn on "Support broken registrar" the phone tries to find the right apartment by guessing. But this guessing will fail when two parties with the same name live in the building.
Q: How can I change settings in-band and out-of-band DTMF signalling?
A: The DTMF type is negotiated during the call setup and cannot be changed manually. Depending on the abilities of the two parties, "out-of-band" is certainly preferable and "in-band" should be used as fallback.
Technical Details: The snom phone (SIP UA) sends the following message inside the INVITE SDP section:
a=fmtp:101 0-16
where
- fmtp is the OOB offer
- 101 example payload ID
- 0-16 defines the range of keys: 0-11 (12) normal keyboard + 5 letters "abcde"
If the receiver accepts OOB the SIP OK response may contain the following in the SDP section:
a=fmtp:101 0-11
If this message is absent, the snom phone assumes in-band" DTMF to be used.
Q: When you talk about “registrations”, does this mean FXS and SIP phone (both hard and soft phones) device registrations or does this include FXO or PSTN gateways as well? Also, what other FXS, FXO devices are integrated?
A: A SIP UA (user agent or terminal) sends out SIP REGISTER messages. The number of registrations pertains to the number of different identifications and locations. E.g., a phone and a Windows Messenger could register as “101” or one phone could register as “103 and ms”, this would be 4 registrations. Gateways can be accessed by a dial plan and do not necessarily need to register.
Q: Can I use two snom phones for VPN without additional equipment?
A: The phones do not have a build-in VPN client. You will need a VPN gateway.
Q: How can I setup snom3xx phones for TLS support?
A: The phone does not listen for TLS connections on any port. The phone only acts as a TLS client for SIPS connections. It setups up and maintains persistent TLS connections to the proxy/registrar for the period of the registration. The "Network identity port" setting is for static SIP udp/tcp port assignment and has no effect on TLS. The "Listen on SIP TCP port" setting is only for TCP.
So basically you dont need any special setting on the phone for using TLS. All you need to do is configure the outbound proxy of your account with a "transport=tls" parameter i.e.:
Registrar: a@b.com Outbound proxy: 202.12.89.23:5061;transport=tls
Q: Ethernet status is showing 'unknown/not connected', what is wrong ?
Ethernet Status: Net Port: Connection Type: unknown Status: not conntected PC Port: Connection Type: unknown Status: not conntected
A: The linux you are using is not supporting this functionality, update the linux !
Audio | Headset | Speaker | Handset | Ringtones
Q: Why is there a humming noise when using the headset?
A: That is a common problem on VoIP phones due to a missing ground connection. The noise is caused by the metal microphone tube and a metal part of the headset, respectively, which act as antennas. There are two simple solutions:
- Connect the phone to the switch with a shielded Ethernet cable + connector (make sure your wall socket/switch/hub has the shielded socket connected to the ground; most of the switches available on the market do).
- In case your network environment is not grounded you can use the phone as a switch and connect the second port of the phone (PC port)to the Ethernet card of the PC by means of a shielded Ethernet cable. The PC card is grounded by default.
Q: Can I use a wireless headset for my snom phone?
A: Yes, you can use GNnetcom GN6210, Plantronics CS60 (DECT, and Plantronics Voyager 5105. They all have PHONE connector on the base station. Connect the headset output of the phone to the PHONE connector on the base station. Use the RJ11 cable that comes with the headsets.
You also have to do some changes in the phone's web interface:
Setup: Line x: TAB SIP: auto answer = ON Preferences: Ringer Device for the headset = Use Headset Type of answering = Use headset (Optional, GNnectom) Advanced: Headset mic volume = 3.
Please note that the headset always has the audio channel open, i.e., you will receive all arriving calls without pressing the button on the headset. GNnetcom has a useful beep before the connection starts, Plantronics does not.
Battery life: ~2 hours
Q: Why is the outgoing audio noisy and low when using the snom headset on the snom300?
A: The headset was originally designed for snom 320/360 but due to design changes on the snom 300 there is a difference in the microphone pin-out which produces the mentioned noise. snom will provide a new headset as soon as possible (aprox. 08/2006). A workaround is to increase the microphone volume via the Web Interface to 7-8.
In any case follow the guidelines mentioned here.
Q: Why do I hear the echo of my own voice on some/all outbound calls?
A: This is a very good description of probable echo causes. If you use Asterisk you should have a look at this explanation on how echo can be avoided at your PSTN gateway.
Q: How can I use the "Music on hold server" functionality?
A: "Music on hold (MOH)" refers to the audio you might want to play to callers while they are waiting after their call was placed on "hold". The MOH Server (can be configured Line/Identity X -> SIP Tab) is an application acting as SIP client which automatically answers to SIP INVITE messages and immediately plays audio from any source or format located anywhere (LAN, Internet). The same syntax as for SIP URIs is used:
Music on hold server: sipaccount@mohserver (e.g. 1000@mymohserver.com)
snom does NOT provide MOH Server applications and has got NO knowledge about currently accesible public MOH servers.
Links:
- MOH Configuration via Web Interface
Q: How do I apply customized ringer melodies and what is the required format?
A: Change the following settings on Line X:
Ringtone: "Custom Melody"
Preferences -> Address Book Ringtones-> Custom Melody URL: HTTP URL pointing to an appropriate wav file, for example:
Custom Melody URL: http://192.168.0.9/test.wav
The wav file itself should be a PCM encoded 8 KHz file at 16bit mono. The time for loading the file should not be longer then 3 seconds ! And the size should be below 250KB.
On a linux based system this format can be obtained from mp3 by typing:
/usr/bin/mpg123 -m -r 8000 -w - -n 190 -q test.mp3 > test.wav
To convert an existing WAV file:
sox GENERIC.wav -c 1 -r 8000 -w SNOM.wav
* The "-c 1" flag makes the output mono.
* The "-r 8000" flag makes the output a 8kHz sample.
* The "-w" flag uses 16 bits ("word") per sample.
When the account configured on line x is called the Wave sound will be audible.
Q: Is it possible to address the internal ringtones via ALERT_INFO ?
A: Yes, 'Alert-Info: <http://127.0.0.1/Bellcore-dr2>' should work as expected. 'Alert-Info: Bellcore-dr2' should also be ok. Bellcore-dr2 to Bellcore-dr5 should work. The melodies may differ from the usual built-in ones.
Update | Factory Reset | Password Loss
Q: The firmware for snom 190/3xx is divided up into several parts. Why is it done in this way and what is the purpose of each part?
A: The idea is to update only the part that needs an update. Therefore, each part has its own flash partition:
- Bootloader "-b-" is the bootloader running a few simple tasks. We do not anticipate frequent changes if any.
- Linux "-l-" contains the linux kernel. Changes will be rare.
- Ramdisk "-r-" is the ramdisk consisting of basic but important files to get the Linux system running (e.g. /dev).
- Application Filesystem (JFFS2) "-j-" is the application named "phone".
Q: After an update to snom3X0-ramdiskToJffs2-br.bin (Rootfs-Version: snom3x0 jffs2 v3.35) or downgrade to snom3x0-jffs2ToRamdisk-br.bin my phone is unusable but showing Ready ... Serial 2.0, what happened?
A: In some cases, this version may lead to the loss of the MAC address during a restart, especially if you did a 'hard' reboot by pulling the power plug. The phone must be returned to snom for repairs.
If you used the above named files for updating/downgrading and your phone is still working fine, please update to the new snom3X0-ramdiskToJffs2-br.bin (Rootfs-Version: snom3x0 jffs2 v3.36). You can then downgrade with the new snom3x0-jffs2ToRamdisk-br.bin, if you like. It is highly recommended that you do not use PoE (power over ethernet) for this update/downgrade session!
Q: After an update to firmware version 4 or above, my phone isn't registering or not receiving calls anymore, why is that ?
A: It may be a matter of compatibility with the proxy server or the other side or a similar reason. Please experiment with the following settings to achieve the highest compatibility; be advised that you may lose safety and maybe some functionality.
Line page SIP tab:
Long SIP-Contact (RFC3840) to "off" Support broken Registrar to "on"
Advanced page:
Filter Packets from Registrar to "off"
Q: How can I set the phone back to admin mode or factory defaults respectively?
A: To set the phone back to admin mode, press the Settings key and confirm the menu item "Administrator Password". On the prompt enter your administrator password, if you have set one, or the default password, which is "0000" (four zeros), and confirm. If neither password returns the phone to admin mode and you need to set the phone back to factory defaults, there are three ways to proceed:
- Via Webinterface: Open your phone's web browser (obtain your phone's IP address by pressing the Help button on your snom 3xx). Choose the URL corresponding to your current firmware version, copy the part following "phonesIpAdr" and paste it after your phone's IP address in the browser's address line. (Or copy the appropriate URL into the address line of your web browser and then change "phonesIPAdr" to your phone's actual IP address.) Then reboot the phone and everything will be accessible again. This will work only when the web interface is not secured by a password.
- Firmware lower than 3.0: http://phonesIpAdr/set_base_en.htm?reset=Reset
- Firmware higher than or equal to 3.0: http://phonesIpAdr/advanced.htm?reset=Reset
- Via Phone GUI:
- snom360: Press the "Settings" Key and scroll down to "Reset Values". Confirm by pressing
!
- snom320: Press the "Settings" Key and press the soft key below the displayed "Reset".
- snom300: Press the "Navigation Button"
"down arrow" and scroll until "Configuration". Press the "Navigation Button" "right arrow" and confirm by pressing
- snom190: Use the arrow keys to navigate to "Configuration". Press the left "softkey" below "Reset".
- snom360: Press the "Settings" Key and scroll down to "Reset Values". Confirm by pressing
- TFTP Update combines the firmware update and the factory reset. It will also work in case of a lost "HTTP" password, i.e., a password securing the phone's web interface.
Q: Why is my phone constantly asking for a password?
A: A workaround may be to switch off the option "Challenge Response on Phone:" on the 'Advanced' page. There is a patch for Asterisk available here that will fix the problem.
Q: Do application firmware updates remove settings that have been previously made?
A: An application update does not remove the settings but an emergency update via a TFTP server does. The phone might also get its values from a settings server.
Display | LEDs | Messages | Keyboard | Appearance
Q: How to control the LEDs?
A: See the VOIP forum page.
Q: What does XML Idle screen description mean?
A: It can be used to change the appearance of the idle screen on a snom360 phone based on the current active outgoing identity. For each SIP line there is a setting "user_xml_screen_url" which can be used like this:
user_xml_screen_url1: http://192.168.0.9/tst.xml
In the above case "identity 1" is used, which causes the screen to change means whenever SIP line 1 is set the current active outgoing identity.
Link: Description and Examples
Q: How to send a text message to the phone in order to appear on the display?
A: There are two different kinds of messages: Desktop messages and SMS (concerning "SMS", see the FAQ question on the VOIP Forum page).
Please follow these steps:
1. Use firmware version 4.3 or higher and the configure the following settings:
Line page SIP tab: * Support broken Registrar to "on" Advanced page: * Filter Packets from Registrar to "off" * Network identity (port): 5060
From Version 6.2.2 onwards you can configure if the text message stays permanently or can be deleted by pressing the "Cancel" key:
Advanced page: * Clear Desktop Message on Cancel: "off" -> message stays permanently * Clear Desktop Message on Cancel: "on" -> message can be deleted by "Cancel" key
2. Re-boot (!!) the phone but do NOT reset it!
3. Set up at least one identity (e.g. 486@mypbx) on the phone and set it up as outgoing identity.
4. Install SIPSAK on 192.168.Y.Y and enter this command:
sipsak -i -M -B "Test" -s sip:486@mypbx
5. "Test" will be displayed on:
* snom360: line above the softkey icons * snom320: 1st line between date and time * snom300: 2nd line
6. Troubleshooting:
- See the SIP trace in order to check if the SIP message has reached the phone e.g.:
MESSAGE sip:486@192.168.X.X:5060;line=gv8x1x75 SIP/2.0 Record-Route: <sip:192.168.Y.Y:7000;ftag=2fcc3043;lr=on> Via: SIP/2.0/UDP 192.168.Y.Y:7000;branch=z9hG4bK2b2.148e3c11.0 Via: SIP/2.0/UDP 192.168.Y.Y:35343;branch=z9hG4bK.29de83da;rport=35344 From: sip:sipsak@192.168.Y.Y:35343;tag=2fcc3043 To: sip:486@ser.intern.snom.de Call-ID: 801910851@192.168.Y.Y CSeq: 1 MESSAGE Content-Type: text/plain Max-Forwards: 16 User-Agent: sipsak 0.9.1 Content-Length: 0 P-hint: usrloc applied Test
- the message is sent to a specific identity on the phone, e.g., sip:486@192.168.X.X and the phone has to be set to that identity before (!!) Only then it can be matched and answers "200 OK" otherwise "404 Not Found".
SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.Y.Y:7000;branch=z9hG4bK2b2.148e3c11.0 Via: SIP/2.0/UDP 192.168.Y.Y:35343;branch=z9hG4bK.29de83da;rport=35344 Record-Route: <sip:192.168.Y.Y:7000;ftag=2fcc3043;lr=on> From: sip:sipsak@192.168.Y.Y:35343;tag=2fcc3043 To: sip:486@mypbx Call-ID: 801910851@192.168.Y.Y CSeq: 1 MESSAGE Content-Length: 0
Note: Starting from version 6.0.5, it is also possible to show special messages for a particular call as long as the Call-Id matches the current call. An example of the MESSAGE method would be:
MESSAGE sip:153@192.168.1.110:3387 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:5060;branch=z9hG4bk212 From: "John" <sip:151@192.168.1.110:5060>;tag=13f53gggg3323 To: "Max" <sip:153@192.168.1.110:3387> Call-ID: 1-5204@192.168.1.110 Cseq: 4 MESSAGE Contact: sip:151@192.168.1.110:5060 Max-Forwards: 70 Alert-Info: Message sound Content-Type: text/plain Content-Length: 34 Invoice id 007
Q: What does the 'Record' keyboard button do?
A: When the 'Record' key is pressed once during a call, the phone sends a SIP INFO message with
Record: on
and a blinking symbol at the display indicates the running recording. Another key press stops the recording, switches the display indicator off, and makes the phone send a SIP INFO message with
Record: off
This default behaviour can be changed via the 'Function Keys' web interface page, see this description also.
Please be aware that the phone is only triggering the start and end of the recording on a remote location which has to perform the recording itself. The phone does not record the voice streams at all.
Settings | Remote Control | Action URL | Mass Deployment
Q: Can the phone post its setting variables with an Action URL?
A: Yes, from firmware 5.0 onwards it is possible to send the following variables of your phone via an Action URL (see also programmable keys):
* usual settings stored on the phone (Settings page) * private settings e.g. passwords are replaced by empty strings * $local for local URI (=own identity replaced at run-time) * $remote for remote URI (=caller ID replaced at run-time) * $call-id for the current call ID (replaced at run-time)
The Action URL can contain variable names starting with $ and "pattern=$variable" strings separated by "&" i.e.
your_pattern1=$variable1&your_pattern2=$variable2&...
Examples:
http://example.invalid/server.php?language=$language http://example.invalid/help.xml?redirect=$redirect_number&time=$redirect_time http://192.168.X.X/test.php?remote=$remote
Action URLs can be assigned to the following events:
DND on/off - Redirection on/off Incoming call/Outgoing call - Setup finished On offhook/onhook - Missed call - On Connected/Disconnected
but those variables which are replaced on run-time do not work with all of them.
The "GET" requests initiated by applying Action URLs are logged and can be easily verified by accessing the Log page e.g.
[2]11/4/2006 10:52:43: Sending post request host = 192.168.X.X:80, file = /test.php?remote=XXX@ser.server.test
Q: Can I control my snom phone remotely?
A: Yes, via the HTTP interface. By means of the HTTP interface you can simulate a person pressing the keys of the phone.
The URL to press a key is the address of your phone with the page 'command.htm' and the post value "key=KEYEVENT". The following KEYEVENTs are known to the firmware of the phone (written exactly like this in capital letters): CANCEL, CLEAR, ENTER, OFFHOOK, ONHOOK, RIGHT, LEFT, FUNCTION, MENU, REDIAL, F1, F2, F3, F4, SPEAKER, DISCONNECT, RECALL, BREAK, 0-9, *, #
Most of the KEYEVENT names are self-explanatory. F1 to F4 are the function keys directly below the display of the phone (please note that the snom 190/200 have no key F4). RECALL is the hold button in the left lower corner of the keypad of the snom 190/200/220.
Example:
http://192.168.0.1/command.htm?key=OFFHOOK
simulates the user of the phone with the IP address 192.168.0.1 picking up the handset.
For the programmable keys use P1-P5 (snom190/200/220) or P1-P12 (snom320/360). For the extension keyboard (snom220/snom360) use prefix "EK", from EK0-EKmax.
The post value "number=1234" dials 1234. In order to place a call from a specific outgoing line (SIP account, identity) attach "&outgoing_uri=user@domain1"
Example:
http://192.168.0.1/command.htm?number=12345678&outgoing_uri=123@domain1
Q: Can I set single phone settings via HTTP requests?
A: Yes. For example, use the below URL (change "phoneIP" to the IP address of your phone) to set the language to German.
http://phoneIP/dummy.htm?settings=save&language=Deutsch
Q: How to trigger the phone to synchronize its settings via mass deployment?
A: The phone can resync the settings with the setting server if it receives a message like:
NOTIFY sip:def@192.168.1.130:2063 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.128:5060 From: <sip:abc@192.168.1.128> To: <sip:def@192.168.1.130:2063> Event: check-sync;reboot=false Date: Tue, 02 Aug 2005 18:25:36 GMT Call-ID: 123456789@192.168.1.130 CSeq: 1300 NOTIFY Contact: <sip:abc@192.168.1.128> Content-Type: application/simple-message-summary Content-Length: 0
With
Event: check-sync;reboot=true
it will reboot first.
Q: What does 'Subscribe Config' mean ?
A: If the setting Subscribe Config is enabled, the phone sends out a SUBSCRIBE message with
Accept: application/x-snom-config
like
SUBSCRIBE sip:111@192.168.0.1:5060;type=user;vendor=snom;product=snom320 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.198:5060;branch=z9hG4bK-zy4mfi3nrnq7;rport From: <sip:111@192.168.0.1:5060>;tag=s0d7ptm3cf To: <sip:111@192.168.0.1:5060;type=user;vendor=snom;product=snom320> Call-ID: 3c267009d467-zjwj968hpi4i@snom320-000413240068 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: <sip:111@192.168.6.198:5060;line=x4kkws5e>;flow-id=1 Event: sip-config Accept: application/x-snom-config Expires: 3600 Content-Length: 0
during boot up and may receive the following example answer message from the registrar:
SIP/2.0 301 Moved Permanently v: SIP/2.0/UDP 192.168.6.198:5060;branch=z9hG4bK-zy4mfi3nrnq7;rport=5060 f: <sip:111@192.168.0.1:5060>;tag=s0d7ptm3cf t: <sip:111@192.168.0.1:5060;type=user;vendor=snom;product=snom320>;tag=nnz854gufw i: 3c267009d467-zjwj968hpi4i@snom320-000413240068 CSeq: 1 SUBSCRIBE m: <http://192.168.0.1:80/config.htm?domain=192.168.0.1&account=111&vendor=snom&product=snom320> l: 0
The phone will then retrieve its usual settings file via the delivered HTTP URL, here
<http://192.168.0.1:80/config.htm?domain=192.168.0.1&account=111&vendor=snom&product=snom320>
Q: Can I provide encrypted user passwords via mass deployment ?
A: Yes, by using the "user_hash" setting. It will be calculated as follows:
user_hash = md5 (user:realm:pass)
where
user = Account realm = Registrar pass = Password
Q: What does "PnP config" on the "Advanced Page" do?
A: When "PNP config" is "ON" (by default) the phone sends a SUBSCRIBE message to a multicast address. All setting servers which have membership to the group can respond to the SUBSCRIBE and send NOTIFY messages with the setting server HTTP URL in the body. The phone then retrieves its settings from the link specified.
This is particularly useful for out of the box setup of devices and mass deployment. The multicast SUBSCRIBE message is sent to 224.0.1.75 and looks like:
SUBSCRIBE sip:MAC%3a000413231323@mycompany.com SIP/2.0 Via: SIP/2.0/UDP 192.168.9.48:5088;rport From: <sip:MAC%3a000413231323@mycompany.com>;tag=1521126922 To: <sip:MAC%3a000413231323@mycompany.com> Call-ID: 1588113540@192.168.9.48 CSeq: 1 SUBSCRIBE Event: ua-profile;profile-type="device";vendor="snom";model="snom360";version="6.2.3" Expires: 0 Accept: application/url Contact: <sip:192.168.9.48:5088> Content-Length: 0
Links:
