snom 360 Legacy Firmware Notes

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This page was created to make the Main Snom 360 Page more usable. - iH

Firmware Notes (Snom 360 firmware 5.3a / 5.3b / 5.4 / 5.5)

Please note what version you are using when you add a bug/note. Also if a new beta fixed one of your bugs, please report it here for us as well. Thank you. - mike240se (3/9/06)

please sign your edits! it is very antisocial to make anonymous edits. - bani

Bugs / Feature Requests

  • MAJOR: Random lockups doing attended transfers while other lines are on hold - bani
    • UPDATE: (Mar13/06) Should be fixed in 5.4. - snomcs

  • MAJOR: (Feb22/06) Phone completely locks up randomly following no pattern, sometimes you hang up the phone and it doesnt hangup and the only thing you can do is unplug it cause its frozen. Random calls are garbled audio and then the phone locks up. This problem can make the phone unusable in a business enviroment. These problems are 5.3, I will test 5.3.6 beta tomorrow. - mike240se
    • UPDATE: (Feb28/06) I had a phone completely lose connectivity today, the computer attached to the phone's switch still was able to access the network, but the phone had an "X" in the top left of the lcd, it could not be pinged or http'd to, the phones buttons still worked, i could go in the menu's but i had to reboot it to get it to work again. 5.3.6, latest linux. - mike240se
    • UPDATE: (Mar6/06) This weekend, two of my phones completely locked up again, couldnt ping or connect to www. I ran ethereal on one of the pc's that was attached to its switch and found that the phone was actually sending continuous arp requests for the mac of the asterisk box, but it just kept ignoring the reply, and the address is was sending the requests from, 192.168.0.106 could not be pinged or http'd to. This is 5.3.6, ulaw, no stun, no srtp. - mike240se
    • UPDATE: (Mar13/06) We identified a problem in LAN with more than 1024 Ethernet devices. There is a workaround available, but this is a really ugly thing. Use it only if you have the same problem. - snomcs
      • UPDATE: (Mar13/06) I got your email, but i only have 10 or less ethernet devices, that includes phones and computers, so this cant be the problem. I am still having the problem, all it does is send arp's, the connectivity is dead, but the gui mostly works. - mike240se

  • FIXED: (Feb23/06) Bad echo while making a call from phone -> asterisk -> teliax via SIP. Other phones dont have any echo when making calls to the pstn via teliax via sip, only the Snom 360's. Both 5.3 and 5.3.6 have this problem. The echo is only heard on the Snom 360 side. Sometimes it goes away a few minutes in, but its pretty bad for the first few minutes of the call. - mike240se
    • NOTE: (Feb28/06) I dont get any echo on teliax with my snom 360. - bani
    • NOTE: (Mar06/06) Echo is generated by the other peer through accoustic or hybrid echo. -hirosh
    • NOTE: (Mar06/06) This is far-end echo. This is a problem if the PSTN gateway. -snomcs
    • NOTE: (Mar13/06) I know where the echo is coming from, but the snom 360 is the only phone that cant deal with it. The rest of the phones using the same gateway dont have this problem, so they are handling it. - mike240se
    • NOTE: (Mar13/06) It is not possible for any phone to handle far-end echo. If you are hearing echo on snom it is likely because of different phone gain settings. - bani
    • NOTE: (Apr15/06) Yeah probably the gains, since its not a problem with the phone, i set it to fixed. - mike240se

  • FIXED: (Mar15/06) The phone doesn't reboot when it receives "Event: reboot" from asterisk 1.2.5. Both challenge_reboot and challenge_checksync are set to on. #asterisk -rx sip notify reboot-snom 100 - hookhook
    • NOTE: (Mar15/06) WHat firmware are you using? I am successfully using the sip notify reboot on 4 snom phones with 5.4 and * 1.2.5 without any problems. Maybe double check your config? - mike240se
    • Re: (Mar15/06) I use 5.4 firmware too. Reboot Command: asterisk -rx "sip notify reboot-snom 100". 100 is the phone number. I use the default sip_notify.conf, it looks like [reboot-snom] Event=>reboot Content-Length=>0 - hookhook
    • NOTE: (Mar15/06) I just re read what you first wrote, challange reboot and challenge sync must be set to OFF! that is your problem :) - mike240se
    • NOTE: (Mar15/06) Thanks Mike! Works perfectly Now. - hookhook

  • MINOR: the dnd_mode!: setting is 180 degrees out of sync with the acutal DND status of the phone. i.e 'dnd_mode!: off' when it is on, and vice versa. A factory reset cures this problem temporarily. This is a major problem if you are using the dnd action urls, to notify your pbx of the phones state, as it toggles the DND status incorrectly. - alan
    • NOTE: This has been fixed in 5.3.6 - snomy

  • MINOR: Phone sends 2 registration requests per second for each account when registration fails. If you have multiple accounts and many phones, this can easily flood an ethernet with incredible amount of traffic when your asterisk server goes down or is misconfigured. - bani
    • NOTE: This appears to have been fixed in 5.3.6 - bani

  • MINOR: Enabling DND while phone is ringing does not stop current ring. It only stops the next ring. - bani

  • MINOR: (Feb/15/06) When pressing volume on the idle screen to adjust ringtone, the ringtone played is 'Ringer 4', not the correct ringtone for the current selected registration. - bani

  • MINOR: (Feb28/06) DTMF someone will need to confirm this but when i use sip-info for dtmf, when we call pagers the pagers get all sorts of extra weird digits or spaces mixed in with what we punched in. Also sometimes automated systems wont respond, but sometimes they do. I use sip only to teliax, no iax, so no jitter buffer. Can someone please confirm this. I will try rfc2833 tonight. I use 5.3.6 beta.- mike240se
    • UPDATE: RFC2833 worked perfect and fixed the problem, it appears sip-info works fine if you are only dialing 1 DTMF digit at a time, if you dial multiple numbers like an account number it doesnt work. I tested it with several bank systems. So it looks like sip info dtmf is broken... Can anyone confirm this? - mike240se
    • NOTE: (Mar13/06) Some gateways advertize INFO and RFC at the same time, and the snom then sends both. INFO is only necessary if the proxies in the middle are interested -snomcs

  • TWEAK: logon_wizard should default to false. - bani

  • TWEAK: challenge_response should default to false. - bani

  • TWEAK: cw_dialtone should default to false. - bani

  • TWEAK: stun_binding_interval[1-12] should default to 300. Right now it defaults to 0.5. - bani

  • TWEAK: When transfering a call, speed dials dont work, only destinations do. So if i assign a button to speed dial 700 for parking, it doesnt work after I press transfer. I cant set it as a destination cause it will keep trying to subscribe to 700 which doesnt have a hint. So if speed dial worked when transfering a call, that would do the trick.

  • FEATURE REQUEST: Extreme overuse of the "huge" font when "small" font is far more appropriate (eg the "reg" menu, directory, menu, settings, redial, caller id, etc) - bani
    • NOTE: Should be a choice, Some of us need that large font to see what the phone is doing. - sdjernes
    • NOTE: The problem with the "huge" font is that it only allows about 18 characters wide. Which means a lot of stuff gets truncated and sometimes useless (eg caller id, directory, redial). Even some of snom's own system menus get truncated! - bani
    • NOTE: This is especially bad when displaying CID Name + Number. Here in NJ, USA cellphones' CID Names often dont help distinguish callers ("New Jersey Wireless Customer"), so w/o the number it makes CID pretty useless for incoming calls, but with the number you lose a lot of the name. --FuriousGeorge

  • FEATURE REQUEST: Option to adjust the ringer volume when you are selecting the ringtone in the menu (Menu->Line->RingTone) - bani

  • FEATURE REQUEST: Option to force the backlight on permanently. - bani

  • FEATURE REQUEST: "Save the current party to the phone book" gives no indication anything was performed. An acknowledgement or confirmation request would be nice. - bani

  • FEATURE REQUEST: USA indications are harsh and/or glitchy. The snom 360 dialtone sounds raspy and aliased, compared to asterisk's Playtones(dial) which sounds exactly like the dialtone I get from my ILEC. snom's busy indication is also completely wrong compared to Playtones(busy). - bani

  • FEATURE REQUEST: (Feb/20/06) Also please fix the call waiting tone for US indications, its loud, intrudes on the active conversation, and the remote party can also hear it! It should be light and subtle and you should be able to talk right through it without having to repeat what you are saying. And the remote party should never hear it. -Mike240se
    • NOTE: (Feb/20/06) I think allowing custom call waiting tones (like custom ringtones) would be a better solution. I am able to talk through it without any problems (audio to remote is not cutoff). I think the reason the remote party hears it (faintly) is because it is so loud in the earpiece that the microphone picks up some of it. - bani
    • NOTE: (Feb21/06) I agree, the main problem is that its like 3 beeps, it should just be 1 light beep. BUT 5.3.6 has an option for the Call Waiting Indicator called "RING" I set it to ring and it did the same as "ON" which means it still gave the same 3 beeps. Is there somewhere to set the cwi ring url in the web interface?? I didnt see it. But this looks like a step in the right direction. - mike240se
    • NOTE: (09May/06) In addition to the comments above, it seems that the Ringer/Ring option on the cwi field does not actually work. 'On' and 'Ringer' do the same thing. A receptionist using a Snom 360 would like her phone to ring for all incoming calls, not just the first call and then beeping for the cwi. I assume the Ringer option would cause the phone to ring. I've installed firmware 5.3.6. Looking at the notes for firmware 6+ it does not seem that any developments have happened here with the exception to the volume of the cwi indicator being turned down as suggested above. Does anyone know when the 'Ring' option will cause the phone to ring on speaker? - cmacmillan

  • FEATURE REQUEST: Option to display digital clock instead of analogue. I know you can make your own idle screen with digital clock through a custom written xml idle page, seems rather silly not to have it as an option in the builtin webserver configuration. maybe provide a selection of pre-made idle pages, like there are pre-made ringtones? - bani

  • FEATURE REQUEST: IAX, ilbc and speex support. - bani

  • FEATURE REQUEST: Holding down keys should repeat (eg volume up/down, scrolling through directory, menus, call history, etc) - bani

  • FEATURE REQUEST: Option to set ringer volume via 'massdeployment' provisioning. - bani

  • FEATURE REQUEST: Redial button works by itself when handset is on-hook, but when off-hook (when screen says "Enter Number") it only works if you push Directory first. It should work the same whether handset is on-hook or off-hook. - bani
    • Note: This appears to have been fixed in 5.3.6. -mike240se 2/21/06

  • FEATURE REQUEST: When you have multiple unsuccessful registrations on the idle screen, it is not possible to tell which line the selector is at (the selector is "hidden"/"invisible" behind unregistered box icons). You can only tell when the selector scrolls past registered lines. a selected 'unregistered' line should have an inverted or 'filled in' box. - bani

  • FEATURE REQUEST: Allow customization of softkey text ("Reg", "Deny",etc) and display text ("Enter Number", "Connected", "DND active!" etc) via 'massdeployment' provisioning, so that end users may localize the menus for their own language or needs (eg chinese, japanese, korean, russian, arabic, etc.). - bani

  • FEATURE REQUEST: (Feb/15/06) Ringtone volume per-line setting. Some custom ringtone are different volume than others, and require different volume settings. Also someone might want e.g. line1 to ring louder than line2. - bani

  • FEATURE REQUEST: (March 15/06) Allow selection of ringer tones 1-10 through the SIP Alert-Info tag. Currently, you can specify Bellcore-dr2 to -dr5, but these are totally different (and less pleasant) than ringer tones 1-10. - mjc

  • FEATURE REQUEST: (March 22/06) An adjustable ethernet MTU setting would be useful, especially in odd situations with VPNs and PPPOE where unexpected packet fragmentation can cause hard-to-isolate errors. - mjc

  • FEATURE REQUEST: (04 March 2006) Allow suppression of 'Missed' call Status Message, on a per-call basis We have an office setup with Asterisk At Home and about 30 SNOM320s. Calls coming in to the prime number will ring several phones, and whoever answers acts as Receptionist. Everybody else in the ring group gets an irrelevant "Missed Call" message. We would like to get a "Missed" call display for one-to-one internal and DID calls, but not for missed Ring Group calls. If SNOM provided a means for selecting whether the Missed Calls status was displayed, via an optional SIP header, we could do the rest in Asterisk using the SIPaddheader() application. - ipaterson
    • Firmware 6.0.0beta now has the option to turn the missed call counter on/off for a specific line, which should be a way to solve your issue. - JustRumours
Image


This page was created to make the Main Snom 360 Page more usable. - iH

Firmware Notes (Snom 360 firmware 5.3a / 5.3b / 5.4 / 5.5)

Please note what version you are using when you add a bug/note. Also if a new beta fixed one of your bugs, please report it here for us as well. Thank you. - mike240se (3/9/06)

please sign your edits! it is very antisocial to make anonymous edits. - bani

Bugs / Feature Requests

  • MAJOR: Random lockups doing attended transfers while other lines are on hold - bani
    • UPDATE: (Mar13/06) Should be fixed in 5.4. - snomcs

  • MAJOR: (Feb22/06) Phone completely locks up randomly following no pattern, sometimes you hang up the phone and it doesnt hangup and the only thing you can do is unplug it cause its frozen. Random calls are garbled audio and then the phone locks up. This problem can make the phone unusable in a business enviroment. These problems are 5.3, I will test 5.3.6 beta tomorrow. - mike240se
    • UPDATE: (Feb28/06) I had a phone completely lose connectivity today, the computer attached to the phone's switch still was able to access the network, but the phone had an "X" in the top left of the lcd, it could not be pinged or http'd to, the phones buttons still worked, i could go in the menu's but i had to reboot it to get it to work again. 5.3.6, latest linux. - mike240se
    • UPDATE: (Mar6/06) This weekend, two of my phones completely locked up again, couldnt ping or connect to www. I ran ethereal on one of the pc's that was attached to its switch and found that the phone was actually sending continuous arp requests for the mac of the asterisk box, but it just kept ignoring the reply, and the address is was sending the requests from, 192.168.0.106 could not be pinged or http'd to. This is 5.3.6, ulaw, no stun, no srtp. - mike240se
    • UPDATE: (Mar13/06) We identified a problem in LAN with more than 1024 Ethernet devices. There is a workaround available, but this is a really ugly thing. Use it only if you have the same problem. - snomcs
      • UPDATE: (Mar13/06) I got your email, but i only have 10 or less ethernet devices, that includes phones and computers, so this cant be the problem. I am still having the problem, all it does is send arp's, the connectivity is dead, but the gui mostly works. - mike240se

  • FIXED: (Feb23/06) Bad echo while making a call from phone -> asterisk -> teliax via SIP. Other phones dont have any echo when making calls to the pstn via teliax via sip, only the Snom 360's. Both 5.3 and 5.3.6 have this problem. The echo is only heard on the Snom 360 side. Sometimes it goes away a few minutes in, but its pretty bad for the first few minutes of the call. - mike240se
    • NOTE: (Feb28/06) I dont get any echo on teliax with my snom 360. - bani
    • NOTE: (Mar06/06) Echo is generated by the other peer through accoustic or hybrid echo. -hirosh
    • NOTE: (Mar06/06) This is far-end echo. This is a problem if the PSTN gateway. -snomcs
    • NOTE: (Mar13/06) I know where the echo is coming from, but the snom 360 is the only phone that cant deal with it. The rest of the phones using the same gateway dont have this problem, so they are handling it. - mike240se
    • NOTE: (Mar13/06) It is not possible for any phone to handle far-end echo. If you are hearing echo on snom it is likely because of different phone gain settings. - bani
    • NOTE: (Apr15/06) Yeah probably the gains, since its not a problem with the phone, i set it to fixed. - mike240se

  • FIXED: (Mar15/06) The phone doesn't reboot when it receives "Event: reboot" from asterisk 1.2.5. Both challenge_reboot and challenge_checksync are set to on. #asterisk -rx sip notify reboot-snom 100 - hookhook
    • NOTE: (Mar15/06) WHat firmware are you using? I am successfully using the sip notify reboot on 4 snom phones with 5.4 and * 1.2.5 without any problems. Maybe double check your config? - mike240se
    • Re: (Mar15/06) I use 5.4 firmware too. Reboot Command: asterisk -rx "sip notify reboot-snom 100". 100 is the phone number. I use the default sip_notify.conf, it looks like [reboot-snom] Event=>reboot Content-Length=>0 - hookhook
    • NOTE: (Mar15/06) I just re read what you first wrote, challange reboot and challenge sync must be set to OFF! that is your problem :) - mike240se
    • NOTE: (Mar15/06) Thanks Mike! Works perfectly Now. - hookhook

  • MINOR: the dnd_mode!: setting is 180 degrees out of sync with the acutal DND status of the phone. i.e 'dnd_mode!: off' when it is on, and vice versa. A factory reset cures this problem temporarily. This is a major problem if you are using the dnd action urls, to notify your pbx of the phones state, as it toggles the DND status incorrectly. - alan
    • NOTE: This has been fixed in 5.3.6 - snomy

  • MINOR: Phone sends 2 registration requests per second for each account when registration fails. If you have multiple accounts and many phones, this can easily flood an ethernet with incredible amount of traffic when your asterisk server goes down or is misconfigured. - bani
    • NOTE: This appears to have been fixed in 5.3.6 - bani

  • MINOR: Enabling DND while phone is ringing does not stop current ring. It only stops the next ring. - bani

  • MINOR: (Feb/15/06) When pressing volume on the idle screen to adjust ringtone, the ringtone played is 'Ringer 4', not the correct ringtone for the current selected registration. - bani

  • MINOR: (Feb28/06) DTMF someone will need to confirm this but when i use sip-info for dtmf, when we call pagers the pagers get all sorts of extra weird digits or spaces mixed in with what we punched in. Also sometimes automated systems wont respond, but sometimes they do. I use sip only to teliax, no iax, so no jitter buffer. Can someone please confirm this. I will try rfc2833 tonight. I use 5.3.6 beta.- mike240se
    • UPDATE: RFC2833 worked perfect and fixed the problem, it appears sip-info works fine if you are only dialing 1 DTMF digit at a time, if you dial multiple numbers like an account number it doesnt work. I tested it with several bank systems. So it looks like sip info dtmf is broken... Can anyone confirm this? - mike240se
    • NOTE: (Mar13/06) Some gateways advertize INFO and RFC at the same time, and the snom then sends both. INFO is only necessary if the proxies in the middle are interested -snomcs

  • TWEAK: logon_wizard should default to false. - bani

  • TWEAK: challenge_response should default to false. - bani

  • TWEAK: cw_dialtone should default to false. - bani

  • TWEAK: stun_binding_interval[1-12] should default to 300. Right now it defaults to 0.5. - bani

  • TWEAK: When transfering a call, speed dials dont work, only destinations do. So if i assign a button to speed dial 700 for parking, it doesnt work after I press transfer. I cant set it as a destination cause it will keep trying to subscribe to 700 which doesnt have a hint. So if speed dial worked when transfering a call, that would do the trick.

  • FEATURE REQUEST: Extreme overuse of the "huge" font when "small" font is far more appropriate (eg the "reg" menu, directory, menu, settings, redial, caller id, etc) - bani
    • NOTE: Should be a choice, Some of us need that large font to see what the phone is doing. - sdjernes
    • NOTE: The problem with the "huge" font is that it only allows about 18 characters wide. Which means a lot of stuff gets truncated and sometimes useless (eg caller id, directory, redial). Even some of snom's own system menus get truncated! - bani
    • NOTE: This is especially bad when displaying CID Name + Number. Here in NJ, USA cellphones' CID Names often dont help distinguish callers ("New Jersey Wireless Customer"), so w/o the number it makes CID pretty useless for incoming calls, but with the number you lose a lot of the name. --FuriousGeorge

  • FEATURE REQUEST: Option to adjust the ringer volume when you are selecting the ringtone in the menu (Menu->Line->RingTone) - bani

  • FEATURE REQUEST: Option to force the backlight on permanently. - bani

  • FEATURE REQUEST: "Save the current party to the phone book" gives no indication anything was performed. An acknowledgement or confirmation request would be nice. - bani

  • FEATURE REQUEST: USA indications are harsh and/or glitchy. The snom 360 dialtone sounds raspy and aliased, compared to asterisk's Playtones(dial) which sounds exactly like the dialtone I get from my ILEC. snom's busy indication is also completely wrong compared to Playtones(busy). - bani

  • FEATURE REQUEST: (Feb/20/06) Also please fix the call waiting tone for US indications, its loud, intrudes on the active conversation, and the remote party can also hear it! It should be light and subtle and you should be able to talk right through it without having to repeat what you are saying. And the remote party should never hear it. -Mike240se
    • NOTE: (Feb/20/06) I think allowing custom call waiting tones (like custom ringtones) would be a better solution. I am able to talk through it without any problems (audio to remote is not cutoff). I think the reason the remote party hears it (faintly) is because it is so loud in the earpiece that the microphone picks up some of it. - bani
    • NOTE: (Feb21/06) I agree, the main problem is that its like 3 beeps, it should just be 1 light beep. BUT 5.3.6 has an option for the Call Waiting Indicator called "RING" I set it to ring and it did the same as "ON" which means it still gave the same 3 beeps. Is there somewhere to set the cwi ring url in the web interface?? I didnt see it. But this looks like a step in the right direction. - mike240se
    • NOTE: (09May/06) In addition to the comments above, it seems that the Ringer/Ring option on the cwi field does not actually work. 'On' and 'Ringer' do the same thing. A receptionist using a Snom 360 would like her phone to ring for all incoming calls, not just the first call and then beeping for the cwi. I assume the Ringer option would cause the phone to ring. I've installed firmware 5.3.6. Looking at the notes for firmware 6+ it does not seem that any developments have happened here with the exception to the volume of the cwi indicator being turned down as suggested above. Does anyone know when the 'Ring' option will cause the phone to ring on speaker? - cmacmillan

  • FEATURE REQUEST: Option to display digital clock instead of analogue. I know you can make your own idle screen with digital clock through a custom written xml idle page, seems rather silly not to have it as an option in the builtin webserver configuration. maybe provide a selection of pre-made idle pages, like there are pre-made ringtones? - bani

  • FEATURE REQUEST: IAX, ilbc and speex support. - bani

  • FEATURE REQUEST: Holding down keys should repeat (eg volume up/down, scrolling through directory, menus, call history, etc) - bani

  • FEATURE REQUEST: Option to set ringer volume via 'massdeployment' provisioning. - bani

  • FEATURE REQUEST: Redial button works by itself when handset is on-hook, but when off-hook (when screen says "Enter Number") it only works if you push Directory first. It should work the same whether handset is on-hook or off-hook. - bani
    • Note: This appears to have been fixed in 5.3.6. -mike240se 2/21/06

  • FEATURE REQUEST: When you have multiple unsuccessful registrations on the idle screen, it is not possible to tell which line the selector is at (the selector is "hidden"/"invisible" behind unregistered box icons). You can only tell when the selector scrolls past registered lines. a selected 'unregistered' line should have an inverted or 'filled in' box. - bani

  • FEATURE REQUEST: Allow customization of softkey text ("Reg", "Deny",etc) and display text ("Enter Number", "Connected", "DND active!" etc) via 'massdeployment' provisioning, so that end users may localize the menus for their own language or needs (eg chinese, japanese, korean, russian, arabic, etc.). - bani

  • FEATURE REQUEST: (Feb/15/06) Ringtone volume per-line setting. Some custom ringtone are different volume than others, and require different volume settings. Also someone might want e.g. line1 to ring louder than line2. - bani

  • FEATURE REQUEST: (March 15/06) Allow selection of ringer tones 1-10 through the SIP Alert-Info tag. Currently, you can specify Bellcore-dr2 to -dr5, but these are totally different (and less pleasant) than ringer tones 1-10. - mjc

  • FEATURE REQUEST: (March 22/06) An adjustable ethernet MTU setting would be useful, especially in odd situations with VPNs and PPPOE where unexpected packet fragmentation can cause hard-to-isolate errors. - mjc

  • FEATURE REQUEST: (04 March 2006) Allow suppression of 'Missed' call Status Message, on a per-call basis We have an office setup with Asterisk At Home and about 30 SNOM320s. Calls coming in to the prime number will ring several phones, and whoever answers acts as Receptionist. Everybody else in the ring group gets an irrelevant "Missed Call" message. We would like to get a "Missed" call display for one-to-one internal and DID calls, but not for missed Ring Group calls. If SNOM provided a means for selecting whether the Missed Calls status was displayed, via an optional SIP header, we could do the rest in Asterisk using the SIPaddheader() application. - ipaterson
    • Firmware 6.0.0beta now has the option to turn the missed call counter on/off for a specific line, which should be a way to solve your issue. - JustRumours
Created by: IronHelix, Last modification: Tue 15 of May, 2012 (18:35 UTC) by admin
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