Re: [asterisk-users] SNOM 360
Hi UxBoD, just create a voicemail for your extension and Asterisk will do the rest!!! Giorgio Incantalupo --[ UxBoD ]-- wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
exten => 123,1,Dial(SIP/123|20) exten => 123,n,Voicemail(u123) would be a start, you can have all kinds of fun... On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
Hmmm, okay. But surely it will just try and ring the extension? Or do you mean setup a seperate extension ie. exten => 1001,1,Dial(sip/1001,20) exten => 1001,2,VoiceMail([EMAIL PROTECTED],u) exten => 1001,3,Hangup() exten => 1001,101,VoiceMail([EMAIL PROTECTED],u) exten => 1001,102,Hangup() exten => 2000,1,VoiceMail([EMAIL PROTECTED],u) exten => 2000,2,HangUp() So on pressing the DND it will send all calls to extention 2000 ? TIA On Fri, 30 Mar 2007 12:57:16 -0400 "Andrew Latham" <[EMAIL PROTECTED]> wrote: > exten => 123,1,Dial(SIP/123|20) > exten => 123,n,Voicemail(u123) > > > would be a start, you can have all kinds of fun... > > > On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote: > > Hi, > > > > I have got my new phone working with Asterisk, and must say it is > > very very good combination. Now I have WMI working, but what I > > would like to be able to do is press the DND button on the phone > > and for all calls to my extension to be forwarded direct to my > > voicemail. > > > > How can this be done please ? > > > > TIA > > > > -- > > This message has been scanned for viruses and dangerous content by > > MailScanner, and is believed to be clean. > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
The snom360 DND button forces the phone to give a 480 do not disturb response. Bails --[ UxBoD ]-- wrote: Hmmm, okay. But surely it will just try and ring the extension? Or do you mean setup a seperate extension ie. exten => 1001,1,Dial(sip/1001,20) exten => 1001,2,VoiceMail([EMAIL PROTECTED],u) exten => 1001,3,Hangup() exten => 1001,101,VoiceMail([EMAIL PROTECTED],u) exten => 1001,102,Hangup() exten => 2000,1,VoiceMail([EMAIL PROTECTED],u) exten => 2000,2,HangUp() So on pressing the DND it will send all calls to extention 2000 ? TIA On Fri, 30 Mar 2007 12:57:16 -0400 "Andrew Latham" <[EMAIL PROTECTED]> wrote: exten => 123,1,Dial(SIP/123|20) exten => 123,n,Voicemail(u123) would be a start, you can have all kinds of fun... On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do is press the DND button on the phone and for all calls to my extension to be forwarded direct to my voicemail. How can this be done please ? TIA -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
Ahhh, awesome. Thank you :) On Fri, 30 Mar 2007 18:47:30 +0100 bails <[EMAIL PROTECTED]> wrote: > The snom360 DND button forces the phone to give a 480 do not disturb > response. > > Bails > --[ UxBoD ]-- wrote: > > Hmmm, okay. But surely it will just try and ring the extension? Or > > do you mean setup a seperate extension ie. > > > > exten => 1001,1,Dial(sip/1001,20) > > exten => 1001,2,VoiceMail([EMAIL PROTECTED],u) > > exten => 1001,3,Hangup() > > exten => 1001,101,VoiceMail([EMAIL PROTECTED],u) > > exten => 1001,102,Hangup() > > exten => 2000,1,VoiceMail([EMAIL PROTECTED],u) > > exten => 2000,2,HangUp() > > > > So on pressing the DND it will send all calls to extention 2000 ? > > > > TIA > > > > On Fri, 30 Mar 2007 12:57:16 -0400 > > "Andrew Latham" <[EMAIL PROTECTED]> wrote: > > > >> exten => 123,1,Dial(SIP/123|20) > >> exten => 123,n,Voicemail(u123) > >> > >> > >> would be a start, you can have all kinds of fun... > >> > >> > >> On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote: > >>> Hi, > >>> > >>> I have got my new phone working with Asterisk, and must say it is > >>> very very good combination. Now I have WMI working, but what I > >>> would like to be able to do is press the DND button on the phone > >>> and for all calls to my extension to be forwarded direct to my > >>> voicemail. > >>> > >>> How can this be done please ? > >>> > >>> TIA > >>> > >>> -- > >>> This message has been scanned for viruses and dangerous content by > >>> MailScanner, and is believed to be clean. > >>> > >>> ___ > >>> --Bandwidth and Colocation provided by Easynews.com -- > >>> > >>> asterisk-users mailing list > >>> To UNSUBSCRIBE or update options visit: > >>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >> > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
bails wrote:
The snom360 DND button forces the phone to give a 480 do not disturb
response.
Bails
--[ UxBoD ]-- wrote:
Hmmm, okay. But surely it will just try and ring the extension? Or do
you mean setup a seperate extension ie.
exten => 1001,1,Dial(sip/1001,20)
exten => 1001,2,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,3,Hangup()
exten => 1001,101,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,102,Hangup()
exten => 2000,1,VoiceMail([EMAIL PROTECTED],u)
exten => 2000,2,HangUp()
Here is a better fix... If extension 1000 is unavailable whether in DND
or just not there... Call rolls over to extension 2000 with the caller
ID "1000 Unavailable" so the person at 2000 will know so and so didn't
answer their phone because 1000 was wasting their life away on youtube.
exten => 1000,1,Dial(SIP/1000|30|tr)
exten => 1000,2,Set(CALLERID(name)="1000 Unavailable")
exten => 1000,3,SayDigits(1000,f)
exten => 1000,4,Playback(vm-isunavail)
exten => 1000,5,Goto(SIP/2000,20|tr)
So say user @ 1000 is named John, you could change the caller ID to
"John UA" (UA short for the obvious (unavailable) as well as the fact
there isn't enough space for the entire string).
exten => 1000,1,Dial(SIP/1000|30|tr)
exten => 1000,2,Set(CALLERID(name)="Transferred Call")
exten => 1000,3,Wait,4
exten => 1000,4,SayDigits(1000,f)
exten => 1000,5,Playback(vm-isunavail)
exten => 1000,6,SIPAddHeader("Alert-Info: ")
exten => 1000,7,Set(CALLERID(name)="John UA")
exten => 1000,8,Dial(SIP/2000|30|tr)
... Works like this... If user John transfers the call... Whoever he
transfers it to will see its a transferred call. If John (extension 1000)
doesn't answer, the obvious occurs. (unavailable)
I currently use this scheme for one client using Snom 320's and 360's.
The caller ID works for most phones I've tested. Polycoms, Aastra's
however, don't expect Aastra's to play the wav file.
--
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net
The happiness of society is the end of government.
John Adams
smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] SNOM 360
Hi Erik, have tried to switch of RTP Encryption on the snom (in the account/identity connected to Asterisk). Cheers Tim On 4/26/07, Erik Wartusch <[EMAIL PROTECTED]> wrote: Hi, I`ve a strange problem since an upgrade from Asterisk 1.2 to 1.4.1. If a SNOM 360 calls (internal) another phone type than SNOM (e.g . Linksys, Thomson we have here), there is no audio transmission anymore. I`ve upgraded the phone to the latest firmware with auto upgrade. No results. With 1.2 it was working well. Any suggestions? Thanks Erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- snom technology AG Tim Koehler Partner Manager [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
Is there a reboot for the phone neccessary? If no then it didn`t work. I tested to call a Linksys phone with deactivated RTP encryption, no audio transmission. Erik Am Donnerstag, 26. April 2007 10:28 schrieb Tim Koehler: > Hi Erik, > > > have tried to switch of RTP Encryption on the snom (in the account/identity > connected to Asterisk). > > Cheers > > Tim > > On 4/26/07, Erik Wartusch <[EMAIL PROTECTED]> wrote: > > Hi, > > > > I`ve a strange problem since an upgrade from Asterisk 1.2 to 1.4.1. > > If a SNOM 360 calls (internal) another phone type than SNOM (e.g > > . Linksys, > > Thomson we have here), there is no audio transmission anymore. > > I`ve upgraded the phone to the latest firmware with auto upgrade. No > > results. > > With 1.2 it was working well. > > Any suggestions? > > > > Thanks > > > > Erik > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users -- === Erik Wartusch Deuromedia Technologies GmbH Barichgasse 40-42 1030 Wien Austria Phone: +43 16986442 1205 Fax: +43 1 6981274 email: [EMAIL PROTECTED] www.deuromedia.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
Him Tim, Tim Koehler wrote: > have tried to switch of RTP Encryption on the snom (in the account/identity > connected to Asterisk). Nice to know you're on the list. :) Grüße, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360
Hi Colin I've been using a Snom 360 for 2 weeks and am generally pleased with it. On Wednesday 11 May 2005 22:12, Colin E. McDonald wrote: > I am having major problems with the first run of Snom 360s that rolled > out last month. > > Issues: > > Speakerphone/Hands Free volume spikes up and down during a call. Haven't seen that problem. > You > have to manually set the volume during every call. When you set the volume, press OK. Then it's stored for next time. > This makes it totally > unusable. The sound will cut out completely at the beginning of a call > sporadically. Have you tried a different provider? > > Call comes through speaker phone after you pick up handset and then cuts > to handset a couple of seconds later I don't have that issue. > > There is a mnaufacturing defect where the display cable is disconnected > so you get what appears to be DOA desk sets. Nor that one. Maybe I was lucky! > > Have to press the retrieve message button twice pretty regularly to get > it to dial vociemail (*97) in asterisk Haven't got the VM button configured yet, or tried to. > > Major problem with calls being dropped when you place callers on hold I haven't tried putting callers on hold yet. Have you updated to the latest firmware? Copy the firmware URL from snom.com into the relevant box on the phone's web interface, save and reboot the phone. Gary -- Gary Stimson Zedcore Systems ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360
Colin Similar to Gary's response in that I haven't seem many of these issues. One that is similar, is that of you saying you need to press voicemail key twice to get *97 (or eqivilent code) This as I understand it is not a fault of snom, but a "feature" of asterisk and the whole MWI "protocol". When asterisk signals the phone to say it has voicemail (any phone) it sends in from an address of "[EMAIL PROTECTED]". the message text is basically that which pops up on the bottom line of the display. When you press the voicemail key, or even the soft voicemail key it first tries to make contact with "unknown" as this helps ensure that the right line acesseses its voicemail without the user having to be aware of which line the voicemail is waiting for them on. You have two choices, a change the address of the MWI indicator to come from "[EMAIL PROTECTED]" on the asterisk box or add some lines in your message-centre context that is similar to exten => Unknown,1,Voicemail etc Either of these will bring asterisk up to the level of the snoms features. I have only one minor issue, and thats if I have several people ringing into the phone, when I am not already on a call (all calls are still in the setup phase) I can't choose by pressing the flashing lights, I have to dump them using the soft "no thanks" or the hard x key You almost sound like you have a earlier firmware issue. The latest one is 3.60f a direct link to the firmware is http://www.snom.com/download/share/ I tell a lie -the very latest firmware is 3.60h - as of the 4th May David On 5/12/05, Gary Stimson <[EMAIL PROTECTED]> wrote: > Hi Colin > > I've been using a Snom 360 for 2 weeks and am generally pleased with it. > > On Wednesday 11 May 2005 22:12, Colin E. McDonald wrote: > > I am having major problems with the first run of Snom 360s that rolled > > out last month. > > > > Issues: > > > > Speakerphone/Hands Free volume spikes up and down during a call. > > Haven't seen that problem. > > > You > > have to manually set the volume during every call. > > When you set the volume, press OK. Then it's stored for next time. > > > This makes it totally > > unusable. The sound will cut out completely at the beginning of a call > > sporadically. > > Have you tried a different provider? > > > > > Call comes through speaker phone after you pick up handset and then cuts > > to handset a couple of seconds later > > I don't have that issue. > > > > > There is a mnaufacturing defect where the display cable is disconnected > > so you get what appears to be DOA desk sets. > > Nor that one. Maybe I was lucky! > > > > > Have to press the retrieve message button twice pretty regularly to get > > it to dial vociemail (*97) in asterisk > > Haven't got the VM button configured yet, or tried to. > > > > > Major problem with calls being dropped when you place callers on hold > > I haven't tried putting callers on hold yet. > > Have you updated to the latest firmware? Copy the firmware URL from snom.com > into the relevant box on the phone's web interface, save and reboot the > phone. > > Gary > > -- > Gary Stimson > Zedcore Systems > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Snom 360
Welcome to VoIP... Your operator needs to take care about QoS when you are doing a download. Alternatively, there are some more-or-less tricky and buggy tricks to stop downloads when you are talking; this needs to be done on your IAD. See for example http://www.voip-info.org/wiki-QoS. CS From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid BenderSent: Wednesday, July 26, 2006 12:46 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Snom 360 Hello List, I am trying to configure QoS for the SNOM 360. I plugged the phone in to the internet and then had the customers computer plug in to the phone. Whith default settings when I talked on the phone it was great. As soon as I started a big download the phone call became unclear. I tried messing around with some settings but to no avail. Anyone have any advice ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SNOM 360
On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: > Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
On 7/28/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote: On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: > Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. As I understand it, you can set a QoS priority if the phone is in a VLAN. When you configure the (Tagged) VLAN, you can specify the priority of the packets in the VLAN. Otherwise, newer firmware allows the setting of TOS values IIRC. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
I am trying to have thier PC run thru the port on the phone and the phone give prioroty to itself and the rest to the PC. When my client does a big download the phone call gets real bad. The docs from SNOM on TOS (or DIFFSERV) is poor and I dont understand it well enough. Anyone have configs or docs on how they did this ? Doid - Original Message - From: "Koopmann, Jan-Peter" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, July 28, 2006 3:17 AM Subject: RE: [asterisk-users] SNOM 360 On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
Also SNOM says by Vlan to set the vlan and then the value for the qos. When you set Vlan to 0 it is supposed to be no Vlan. However once I set it the vlan on the SNOM to 0 and I reboot the phone is no long accessable from the network and I have to reset it. Dovid - Original Message - From: "Koopmann, Jan-Peter" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, July 28, 2006 3:17 AM Subject: RE: [asterisk-users] SNOM 360 On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote: Does anyone know how to set up QoS on the SNOM 360 ? Thanks. What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a Snom 360 that will manage things for you. AFAIK all you can do is tell the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of the equipment managing the bottleneck (firewall, router whatever) to use this information and manage your traffic accordingly. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
I would be surprised if the problem is at the phone. I have nearly a hundred 360s, 190s and not one of them suffers from that problem in the default setting. The phone handles it automatically. BUT..if I download from an external site and I pipe the call over the internet without setting any traffic shaping on the router then it gets jumpy. Also, you may experience the same problem if you're somehow saturating the network interface on the switch or the asterisk server (both which is highly unlikely). Check you have some sort of traffic shaping on your router and ensure you have a decent switch. I like m0n0wall for routers and cisco for switches. > -- > > Message: 9 > Date: Fri, 28 Jul 2006 09:08:17 -0400 > From: "Dovid Bender" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] SNOM 360 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; format=flowed; charset="Windows-1252"; > reply-type=original > > I am trying to have thier PC run thru the port on the phone and the phone > give prioroty to itself and the rest to the PC. When my client does a big > download the phone call gets real bad. The docs from SNOM on TOS (or > DIFFSERV) is poor and I dont understand it well enough. Anyone have configs > or docs on how they did this ? > > Doid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SNOM 360
On Friday, July 28, 2006 3:08 PM Dovid Bender wrote: > I am trying to have thier PC run thru the port on the phone and the > phone give prioroty to itself and the rest to the PC. When my client > does a big download the phone call gets real bad. The docs from SNOM > on TOS (or DIFFSERV) is poor and I dont understand it well enough. > Anyone have configs or docs on how they did this ? I would be surprised to learn that the Snom is actively doing traffic management itself. Traffic managment must be done at the bottleneck to be halfway successful. Let's assume you are doing a download and you snom would do traffic management giving itself priority. What if your co-worker is doing a huge download? How should your snom know and throttle his download? No way. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
On 7/28/06, Dovid Bender <[EMAIL PROTECTED]> wrote: Also SNOM says by Vlan to set the vlan and then the value for the qos. When you set Vlan to 0 it is supposed to be no Vlan. However once I set it the vlan on the SNOM to 0 and I reboot the phone is no long accessable from the network and I have to reset it. The "Qos" field is part of the 802.11q header, so is only available if a VLAN has been configured. VLAN 0 is a perfectly valid VLAN, and will cause an 802.11q packet header to be added to all the phone traffic. This will then only work if the rest of the network understands "tagged" VLAN 0 packets. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote: On Friday, July 28, 2006 3:08 PM Dovid Bender wrote: > I am trying to have thier PC run thru the port on the phone and the > phone give prioroty to itself and the rest to the PC. When my client > does a big download the phone call gets real bad. The docs from SNOM > on TOS (or DIFFSERV) is poor and I dont understand it well enough. > Anyone have configs or docs on how they did this ? I would be surprised to learn that the Snom is actively doing traffic management itself. Traffic managment must be done at the bottleneck to be halfway successful. Let's assume you are doing a download and you snom would do traffic management giving itself priority. What if your co-worker is doing a huge download? How should your snom know and throttle his download? No way. That is a different problem entirely, and as you say, the snom cannot do anything about a remote bottleneck (except perhaps theough QoS and TOS flags in the data it sends). The snom does seem to manage its two local ports properly though but this cannot be hard. Worst case is that the snom needs about 128Kb/s - Not hard on a 100Mb/s full duplex connection :) Dovid - Have you identified where the bottleneck is in this case? You do not specify as far as I can see. Is the VoIP call using the internet, or is it local? Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
- Original Message - From: "Steve Davies" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, July 31, 2006 6:01 AM Subject: Re: [asterisk-users] SNOM 360 On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote: On Friday, July 28, 2006 3:08 PM Dovid Bender wrote: > I am trying to have thier PC run thru the port on the phone and the > phone give prioroty to itself and the rest to the PC. When my client > does a big download the phone call gets real bad. The docs from SNOM > on TOS (or DIFFSERV) is poor and I dont understand it well enough. > Anyone have configs or docs on how they did this ? I would be surprised to learn that the Snom is actively doing traffic management itself. Traffic managment must be done at the bottleneck to be halfway successful. Let's assume you are doing a download and you snom would do traffic management giving itself priority. What if your co-worker is doing a huge download? How should your snom know and throttle his download? No way. That is a different problem entirely, and as you say, the snom cannot do anything about a remote bottleneck (except perhaps theough QoS and TOS flags in the data it sends). The snom does seem to manage its two local ports properly though but this cannot be hard. Worst case is that the snom needs about 128Kb/s - Not hard on a 100Mb/s full duplex connection :) Dovid - Have you identified where the bottleneck is in this case? You do not specify as far as I can see. Is the VoIP call using the internet, or is it local? Regards, Steve It is using the internet. The problem is when a user starts a big download. The phone call goes to s***. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
- Original Message - From: "Steve Davies" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, July 31, 2006 5:57 AM Subject: Re: [asterisk-users] SNOM 360 On 7/28/06, Dovid Bender <[EMAIL PROTECTED]> wrote: Also SNOM says by Vlan to set the vlan and then the value for the qos. When you set Vlan to 0 it is supposed to be no Vlan. However once I set it the vlan on the SNOM to 0 and I reboot the phone is no long accessable from the network and I have to reset it. The "Qos" field is part of the 802.11q header, so is only available if a VLAN has been configured. VLAN 0 is a perfectly valid VLAN, and will cause an 802.11q packet header to be added to all the phone traffic. This will then only work if the rest of the network understands "tagged" VLAN 0 packets. Regards, Steve When I set it to 0 I loose all conectivity to the phone. Cant ping it or anything. I have to reset it from the phone to get access to it again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
On 8/4/06, Dovid Bender <[EMAIL PROTECTED]> wrote: > > The snom does seem to manage its two local ports properly though but > this cannot be hard. Worst case is that the snom needs about 128Kb/s - > Not hard on a 100Mb/s full duplex connection :) > > Dovid - Have you identified where the bottleneck is in this case? You > do not specify as far as I can see. Is the VoIP call using the > internet, or is it local? > It is using the internet. The problem is when a user starts a big download. The phone call goes to s***. Dovid, There are devices on the market that claim to prioritise certain traffic in favour of downloads and browsing. This can obviously only prioritise traffic outbound, but on an ADSL link, this is often what is needed. Try googling for such a device? We don't use them ourselves as we operate on the principle that a "free" call is worth what you pay for it :) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
On 8/4/06, Dovid Bender <[EMAIL PROTECTED]> wrote: > > The "Qos" field is part of the 802.11q header, so is only available if > a VLAN has been configured. > > VLAN 0 is a perfectly valid VLAN, and will cause an 802.11q packet > header to be added to all the phone traffic. This will then only work > if the rest of the network understands "tagged" VLAN 0 packets. > When I set it to 0 I loose all conectivity to the phone. Cant ping it or anything. I have to reset it from the phone to get access to it again. Of course you do - nothing else on your network is in VLAN 0, so you lose connectivity. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360
Dovid Bender wrote: - Original Message - From: "Steve Davies" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, July 31, 2006 6:01 AM Subject: Re: [asterisk-users] SNOM 360 On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote: On Friday, July 28, 2006 3:08 PM Dovid Bender wrote: > I am trying to have thier PC run thru the port on the phone and the > phone give prioroty to itself and the rest to the PC. When my client > does a big download the phone call gets real bad. The docs from SNOM > on TOS (or DIFFSERV) is poor and I dont understand it well enough. > Anyone have configs or docs on how they did this ? I would be surprised to learn that the Snom is actively doing traffic management itself. Traffic managment must be done at the bottleneck to be halfway successful. Let's assume you are doing a download and you snom would do traffic management giving itself priority. What if your co-worker is doing a huge download? How should your snom know and throttle his download? No way. That is a different problem entirely, and as you say, the snom cannot do anything about a remote bottleneck (except perhaps theough QoS and TOS flags in the data it sends). The snom does seem to manage its two local ports properly though but this cannot be hard. Worst case is that the snom needs about 128Kb/s - Not hard on a 100Mb/s full duplex connection :) Dovid - Have you identified where the bottleneck is in this case? You do not specify as far as I can see. Is the VoIP call using the internet, or is it local? Regards, Steve It is using the internet. The problem is when a user starts a big download. The phone call goes to s***. Dovid, I would guess that: First thing would be quick&dirty ASCII drawing, showing where is the PC, the SNOM and the "sources/destinations" of the Internet and VOIP traffic. You mentioned download, assuming this is a DSL connection, this would be, when it arrive at the IP phone, would be too late to do anything, IF you are bumping into a bottleneck in the DSL downstream. What 'direction' of the voice path is suffering, did you capture the traffic (is it suffering because of jitter, packet loss, ...) ? Like others mention, QoS (the buzzword :-), is a very wide and generic term, and you will need to 'isolate' the problem to see if a solution is feasible. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 problem
Did you tried to turn off te option "Filter Packets from Registrar" on the Advanced page? That is the only case, when a snom sends a 403 (= Forbidden). Regards Nils Ohlmeier On Friday 02 September 2005 09:11, altus wrote: > 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 > 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" > ;tag=as3405ec0a" > But its 301 calling the snom360(user 310)??? > BUT > If I change the phone's ip and tell it to connect to eth0,not eth0:1 it > works,same account settings same everything? > The snom190 worked this way > Any Ideas why? -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Opinions
I'm not too pleased with the phones, I have about 40 of them, some of the displays tend to die and the dial pad feels to 'mushy' IMHO, just like the keys on a good old ZX80 computer Also I'm having some issues with sound quality on some phones, but I still need to switch some phones to see if that is really an issue of the phone. Also if you want to use * call files, with the 360 you will run into a big where the call is being redialled as if it failed while in fact the call is ongoing. Annoying and haven't found out if that is an * bug or Snom bug. The Snom 190's do not have this problem. Just my $0.02 (which is really not a lot these days!) :) On Sat, 12 Nov 2005, Curren C. Calhoun wrote: I¹m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am open to any other suggestions as well... Thanks Curren ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Opinions
Cleaner looking, 12 line apperances, affordable sidecar, runs on linux, developing XML services, very programable, buttons are firm in a good way, simple layout for users installing 30 right now... On 11/12/05, Remco Barende <[EMAIL PROTECTED]> wrote: > I'm not too pleased with the phones, I have about 40 of them, some of the > displays tend to die and the dial pad feels to 'mushy' IMHO, just like the > keys on a good old ZX80 computer > > Also I'm having some issues with sound quality on some phones, but I still > need to switch some phones to see if that is really an issue of the phone. > > Also if you want to use * call files, with the 360 you will run into a big > where the call is being redialled as if it failed while in fact the call > is ongoing. Annoying and haven't found out if that is an * bug or Snom > bug. The Snom 190's do not have this problem. > > Just my $0.02 (which is really not a lot these days!) :) > > On Sat, 12 Nov 2005, Curren C. Calhoun wrote: > > > I¹m looking to add in some Snom 360 phones, could anyone give thoughts or > > opinions about the speakerphone, general quality... Also the phone would > > need to be powered over Ethernet... > > > > I like some of the listed features and the expandability of the phone but am > > open to any other suggestions as well... > > > > Thanks > > > > > > Curren > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Opinions
Thanks that's the type of info I'm looking for... I've heard some early grumblings but wanted to see if anything else has come up... > From: Remco Barende <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET) > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Snom 360 Opinions > > I'm not too pleased with the phones, I have about 40 of them, some of the > displays tend to die and the dial pad feels to 'mushy' IMHO, just like the > keys on a good old ZX80 computer > > Also I'm having some issues with sound quality on some phones, but I still > need to switch some phones to see if that is really an issue of the phone. > > Also if you want to use * call files, with the 360 you will run into a big > where the call is being redialled as if it failed while in fact the call > is ongoing. Annoying and haven't found out if that is an * bug or Snom > bug. The Snom 190's do not have this problem. > > Just my $0.02 (which is really not a lot these days!) :) > > On Sat, 12 Nov 2005, Curren C. Calhoun wrote: > >> I¹m looking to add in some Snom 360 phones, could anyone give thoughts or >> opinions about the speakerphone, general quality... Also the phone would >> need to be powered over Ethernet... >> >> I like some of the listed features and the expandability of the phone but am >> open to any other suggestions as well... >> >> Thanks >> >> >> Curren >> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Opinions
Curren, Can you tell us a little more about the environment you are deploying these phones in, how many phones and what kind of setup? Omar A. SabekOn 11/12/05, Curren C. Calhoun <[EMAIL PROTECTED]> wrote: Thanks that's the type of info I'm looking for...I've heard some early grumblings but wanted to see if anything else has comeup...> From: Remco Barende < [EMAIL PROTECTED]>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion> <asterisk-users@lists.digium.com>> Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET) > To: Asterisk Users Mailing List - Non-Commercial Discussion> <asterisk-users@lists.digium.com>> Subject: Re: [Asterisk-Users] Snom 360 Opinions >> I'm not too pleased with the phones, I have about 40 of them, some of the> displays tend to die and the dial pad feels to 'mushy' IMHO, just like the> keys on a good old ZX80 computer> > Also I'm having some issues with sound quality on some phones, but I still> need to switch some phones to see if that is really an issue of the phone.>> Also if you want to use * call files, with the 360 you will run into a big > where the call is being redialled as if it failed while in fact the call> is ongoing. Annoying and haven't found out if that is an * bug or Snom> bug. The Snom 190's do not have this problem.> > Just my $0.02 (which is really not a lot these days!) :)>> On Sat, 12 Nov 2005, Curren C. Calhoun wrote:>>> I¹m looking to add in some Snom 360 phones, could anyone give thoughts or >> opinions about the speakerphone, general quality... Also the phone would>> need to be powered over Ethernet...>>>> I like some of the listed features and the expandability of the phone but am >> open to any other suggestions as well...>>>> Thanks>>>>>> Curren>>___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 360 problems
> 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 only on the Snom. When using > a Zap line or to another sip phone. I've tweaked the * for echo and > managed to only create echo and piss everyone else off, pounded the > settings in the Snom trying to find anything, and updated the firmware > to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 > v3.36 after noticing a changelog that sounded like it may have related > to echo. Not even a slight reduction in echo so far. > > A second serious problem is Call join. Even with "Call join on Xfer (2 > calls)" OFF if the user is doing a transfer of one call when a second > starts ringing the 2 callers get bridged, no transfer. Really nice, now > I have two customers talking to each other with no clue what's going on > and neither gets who they were trying to reach. > > Any ideas on what I can try next? This firmware works well for us: snom360-SIP 4.1 available here: http://snom.com/download/share/snom360-4.1-SIP-j.bin No echo and overall voice quality is excellent. Did you check the codecs on the snom and on asterisk (sip.conf)? Is Silence Suppression off on the snom? If you would post your config (under settings on the snom) we could have a closer look in the problem. Regards, Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 problems
Guido Hecken wrote: 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 only on the Snom. When using a Zap line or to another sip phone. I've tweaked the * for echo and managed to only create echo for everyone else, pounded the settings in the Snom trying to find anything, and updated the firmware to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 v3.36 after noticing a changelog that sounded like it may have related to echo. Not even a slight reduction in echo so far. A second serious problem is Call join. Even with "Call join on Xfer (2 calls)" OFF if the user is doing a transfer of one call when a second starts ringing the 2 callers get bridged, no transfer. Really nice, now I have two customers talking to each other with no clue what's going on and neither gets who they were trying to reach. Any ideas on what I can try next? This firmware works well for us: snom360-SIP 4.1 available here: http://snom.com/download/share/snom360-4.1-SIP-j.bin No echo and overall voice quality is excellent. Did you check the codecs on the snom and on asterisk (sip.conf)? Is Silence Suppression off on the snom? If you would post your config (under settings on the snom) we could have a closer look in the problem. I believe I'll try the 5.5.1b firmware as suggested in another response for the second problem. yes, yes and attached. Thanks for your help. language!: English redirect_number!: redirect_busy_number!: redirect_event!: none redirect_time!: phone_type!: codec_tos!: 160 mac&: 000413231FA6 setting_server!: subscribe_config!: off ip_adr!: 10.11.10.100 netmask!: 255.255.0.0 update_server!: dns_domain!: cincinnati dns_server1!: 10.11.0.1 dns_server2!: dhcp!: off gateway!: 10.11.0.1 phone_name!: utc_offset!: -18000 ntp_server!: 10.12.0.2 lcserver1!: ring_sound!: Ringer4 http_proxy!: http_port!: 80 http_user!: http_pass!: http_scheme!: off https_port!: 443 webserver_type!: http_https webserver_cert!: dst!: 3600 04.01.07 02:00:00 10.05.07 02:00:00 timezone!: USA-5 contrast!: 16 sip_retry_t1!: 500 session_timer!: 3600 network_id_port!: max_forwards!: 70 user_phone!: off active_line!: 1 outgoing_identity!: 1 challenge_response!: on refer_brackets!: off sip_proxy!: register_http_contact!: off cmc_feature!: off filter_registrar!: off challenge_reboot!: off challenge_checksync!: off action_dnd_on_url!: action_dnd_off_url!: action_redirection_on_url!: action_redirection_off_url!: action_incoming_url!: action_outgoing_url!: action_setup_url!: action_offhook_url!: action_onhook_url!: action_missed_url!: action_connected_url!: action_disconnected_url!: aoc_amount_display!: off aoc_pulse_currency!: $ aoc_cost_pulse!: 1 rtp_port_start!: 49152 rtp_port_end!: 65534 preselection_nr!: auto_dial!: 5 dtmf_payload_type!: 101 dnd_mode!: on privacy_in!: off privacy_out!: off admin_mode_login!: admin_mode_password!: admin_mode_password_confirm!: admin_mode!: on tone_scheme!: USA vol_speaker!: 1 vol_ringer!: 7 vol_handset!: 15 vol_headset!: 10 vol_speaker_mic!: 0 vol_handset_mic!: 1 vol_headset_mic!: 0 log_level!: 5 auto_connect_type!: auto_connect_type_handsfree auto_connect_indication!: on logon_wizard!: on guess_number!: on guess_start_length!: 4 friends_ring_sound!: Ringer4 family_ring_sound!: Ringer2 colleagues_ring_sound!: Ringer6 vip_ring_sound!: Ringer4 break_key!: false publish_presence!: off edit_alpha_mode!: 123 display_method!: display_name call_waiting!: on cw_dialtone!: on disable_speaker!: off no_dnd!: off mute!: off dirty_host_ttl!: headset_device!: none update_policy!: never_update conf_hangup!: on enum_suffix!: e164.arpa mwi_notification!: silent vlan!: vlan_id!: vlan_qos!: block_url_dialing!: off release_sound!: off deny_all_feature!: off transfer_on_hangup!: on ethernet_replug!: nothing mwi_dialtone!: stutter support_idna!: off custom_melody_url!: ringer_headset_device!: speaker dtmf_speaker_phone!: off presence_timeout!: 15 require_prack!: off offer_gruu!: on offer_mpo!: off firmware_status!: firmware_interval!: firmware!: http://snom.com/download/snom360-ramdiskToJffs2-3.36-br.bin bootloader!: update_filename!: update_host_b!: update_host_f!: sip_port!: 2051 web_language!: English call_completion!: off callpickup_dialoginfo!: on use_backlight!: on reset_settings!: date_us_format!: on time_24_format!: off call_join_xfer!: off alert_info_playback!: on ringing_time!: 60 silence_compression!: off syslog_server!: screen_saver_timeout!: 60 intercom_enabled!: off with_flash!: on snmp_trusted_addresses!: snmp_port!: 161 short_form!: off audio_device_indicator!: on license_data&: Mac:000413231F
Re: [Asterisk-Users] Snom 360 problems
On Fri, 24 Mar 2006, Brian Kennedy wrote: Anyone have a Snom they're happy with? How did you manage that? :) I would be happier if snom fixed the US indications and the giant 3000 point font they use for everything. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 problems
I may need a consultant that can help with some problems. We had 2 smaller satellite offices on the same asterisk systems with no trouble. We've just upgraded the main office and hit troubles. There's no going back because we've entirely outgrown our old system. Any * consultants around Cincinnati want a peek? Brian Kennedy wrote: 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 only on the Snom. When using a Zap line or to another sip phone. I've tweaked the * for echo and managed to only create echo and piss everyone else off, pounded the settings in the Snom trying to find anything, and updated the firmware to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 v3.36 after noticing a changelog that sounded like it may have related to echo. Not even a slight reduction in echo so far. A second serious problem is Call join. Even with "Call join on Xfer (2 calls)" OFF if the user is doing a transfer of one call when a second starts ringing the 2 callers get bridged, no transfer. Really nice, now I have two customers talking to each other with no clue what's going on and neither gets who they were trying to reach. Any ideas on what I can try next? Thanks... ...Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] snom 360 auto answer
This is my code (that I copied form somewhere) for paging a group of
phones. By dialling 99 it will page phones 2101, 2102 and 2105.
Just include the context ext-paging in your dial plan and modify the
extension numbers and all should be good.
This works on Linksys Phones but should also work on Snoms.
I hope this helps you.
[ext-paging]
exten => PAGE2101,1,GotoIf($[ ${CALLERID(number)} = 2101 ]?skipself)
exten => PAGE2101,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten => PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
exten => PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten => PAGE2101,n,Dial(SIP/2101,5)
exten => PAGE2101,n(skipself),Noop(Not paging originator)
exten => PAGE2102,1,GotoIf($[ ${CALLERID(number)} = 2102 ]?skipself)
exten => PAGE2102,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten => PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
exten => PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten => PAGE2102,n,Dial(SIP/2102,5)
exten => PAGE2102,n(skipself),Noop(Not paging originator)
exten => PAGE2105,1,GotoIf($[ ${CALLERID(number)} = 2105 ]?skipself)
exten => PAGE2105,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten => PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
exten => PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten => PAGE2105,n,Dial(SIP/2105,5)
exten => PAGE2105,n(skipself),Noop(Not paging originator)
exten => Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]
aging)
exten =>
99,1,Page(LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/PAGE
[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten => _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten => _99,n,SIPAddHeader(Call-Info:
\;answer-after=0)
;exten => _99,n,Dial(SIP/${EXTEN:2})
exten => _99,1,Set(__SIPADDHEADER=Call-Info:
answer-after=0)
exten =>
_99,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten => _99,n,Set(__ALERT_INFO=Ring Answer)
exten => _99,n,Dial(SIP/${EXTEN:2})
__
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RE: [asterisk-users] snom 360 auto answer
Thankyou David,
It works for Linksys,but not for snom 360.
Do I need to change someting using web UI ?
--- "Klaverstyn, David C"
<[EMAIL PROTECTED]> wrote:
> This is my code (that I copied form somewhere) for
> paging a group of
> phones. By dialling 99 it will page phones 2101,
> 2102 and 2105.
>
>
>
> Just include the context ext-paging in your dial
> plan and modify the
> extension numbers and all should be good.
>
>
>
> This works on Linksys Phones but should also work on
> Snoms.
>
>
>
> I hope this helps you.
>
>
>
>
>
> [ext-paging]
>
> exten => PAGE2101,1,GotoIf($[ ${CALLERID(number)} =
> 2101 ]?skipself)
>
> exten => PAGE2101,n,Set(__SIPADDHEADER=Call-Info:
> \;answer-after=0)
>
> exten => PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
>
> exten =>
> PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
>
> exten => PAGE2101,n,Dial(SIP/2101,5)
>
> exten => PAGE2101,n(skipself),Noop(Not paging
> originator)
>
>
>
> exten => PAGE2102,1,GotoIf($[ ${CALLERID(number)} =
> 2102 ]?skipself)
>
> exten => PAGE2102,n,Set(__SIPADDHEADER=Call-Info:
> \;answer-after=0)
>
> exten => PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
>
> exten =>
> PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
>
> exten => PAGE2102,n,Dial(SIP/2102,5)
>
> exten => PAGE2102,n(skipself),Noop(Not paging
> originator)
>
>
>
> exten => PAGE2105,1,GotoIf($[ ${CALLERID(number)} =
> 2105 ]?skipself)
>
> exten => PAGE2105,n,Set(__SIPADDHEADER=Call-Info:
> \;answer-after=0)
>
> exten => PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
>
> exten =>
> PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
>
> exten => PAGE2105,n,Dial(SIP/2105,5)
>
> exten => PAGE2105,n(skipself),Noop(Not paging
> originator)
>
>
>
>
>
> exten => Debug,1,Noop(dialstr is
>
LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]
> aging)
>
> exten =>
>
99,1,Page(LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/PAGE
> [EMAIL PROTECTED])
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Jason Kim
> Sent: Monday, 8 January 2007 2:30 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] snom 360 auto answer
>
>
>
> Hi,
>
>
>
> I'm testing paging using snom 360.
>
> Can someone correct my dialplan?
>
>
>
> Regards,
>
> Jason.
>
>
>
> ==
>
> ;exten => _99,1,SIPAddHeader(Call-Info:
>
> Answer-After=0)
>
> ;exten => _99,n,SIPAddHeader(Call-Info:
>
> \;answer-after=0)
>
> ;exten => _99,n,Dial(SIP/${EXTEN:2})
>
>
>
> exten => _99,1,Set(__SIPADDHEADER=Call-Info:
>
> answer-after=0)
>
> exten =>
>
> _99,n,Set(__SIP_URI_OPTIONS=intercom=true)
>
> exten => _99,n,Set(__ALERT_INFO=Ring Answer)
>
> exten => _99,n,Dial(SIP/${EXTEN:2})
>
>
>
>
>
> __
>
> Do You Yahoo!?
>
> Tired of spam? Yahoo! Mail has the best spam
> protection around
>
> http://mail.yahoo.com
>
> ___
>
> --Bandwidth and Colocation provided by Easynews.com
> --
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>
>
> asterisk-users mailing list
>
> To UNSUBSCRIBE or update options visit:
>
>
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
> > ___
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>
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Re: [asterisk-users] SNOM 360 Rejecting Calls
Hi, Is this a real question? I'm asking because it seems to be a babelfish translation of a post in a german forum http://www.ip-phone-forum.de/archive/index.php/t-99696.html from over a year ago. (is their date wrong?) Dovid B wrote: > Does someone know coincidentally the cause for the error message specified > in the Subject? Probably misconfiguration. ;) > 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 registers itself as expected, and outgoing telephone calls can > be led problem-free. Detailed telephone calls however do not function (a call > is signaled, which rings then however after 3 time on the mailbox is sent). > > The SIP log shows that the telephone sees the INVITE of the Registrar, it > however in principle with "486 Busy here" answered (that is then also the > reason, why the detailed call is sent on the mailbox). The message mentioned > "Denying call appears contemporaneous id=X reason=unconditional" in the log, > whereby X was so far always a negative, one-digit number. > > Does someone have an idea? Sounds like a bug, I'd suggest filing a bug report on both the Snom and the Digium issue trackers with severity set to major (since you can't make any calls at all!). ;) No, seriously now: I would suspect you have activated some kind of call forwarding rule or DND on the Snom (have a look at the prefs.htm page on the built in web server). Which version of the firmware are you using? Did you try to reset the phone to it's factory defaults? Does ist work without NAT? How does your sip.conf and extensions.conf look like? What's the output of asterisk -vvvr ? Regards Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM 360 Rejecting Calls
I was super super tired when I sent it. I had the same problem as the posted from http://www.ip-phone-forum.de/archive/index.php/t-99696.html and he had no response to his issue. So I reposted here. Resetting the phone took care of the issue. After being up for 4 days straight you have virtually no brain cells left. Thanks for the idea to rest it (don't know why I didn't think of that). - Original Message - From: "Philipp Kempgen" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, May 11, 2007 2:19 AM Subject: Re: [asterisk-users] SNOM 360 Rejecting Calls Hi, Is this a real question? I'm asking because it seems to be a babelfish translation of a post in a german forum http://www.ip-phone-forum.de/archive/index.php/t-99696.html from over a year ago. (is their date wrong?) Dovid B wrote: Does someone know coincidentally the cause for the error message specified in the Subject? Probably misconfiguration. ;) 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 registers itself as expected, and outgoing telephone calls can be led problem-free. Detailed telephone calls however do not function (a call is signaled, which rings then however after 3 time on the mailbox is sent). The SIP log shows that the telephone sees the INVITE of the Registrar, it however in principle with "486 Busy here" answered (that is then also the reason, why the detailed call is sent on the mailbox). The message mentioned "Denying call appears contemporaneous id=X reason=unconditional" in the log, whereby X was so far always a negative, one-digit number. Does someone have an idea? Sounds like a bug, I'd suggest filing a bug report on both the Snom and the Digium issue trackers with severity set to major (since you can't make any calls at all!). ;) No, seriously now: I would suspect you have activated some kind of call forwarding rule or DND on the Snom (have a look at the prefs.htm page on the built in web server). Which version of the firmware are you using? Did you try to reset the phone to it's factory defaults? Does ist work without NAT? How does your sip.conf and extensions.conf look like? What's the output of asterisk -vvvr ? Regards Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 extension lights
Ross Kevlin wrote: > 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? > There is a page on the wiki that subscribes this. If you have a proper subscription and you have the hints, it "should" work as we usually say in this business when we don't know what the problem is. If you are using CVS head, you can limit the context reachable for subscriptions with the "subscribecontext" setting. I haven't tried with the 360 yet, so I might have to power up that phone. For those of you that haven't explored the subscription support in the Asterisk SIP channel: The key to get device state notification in SIP subscriptions and the AMI, the Asterisk manager interface, is the use of a "hint" priority. The phone subscribe to an extension, but in order for the PBX to know which phone that is connected to an extension, you need to tell Asterisk what relationship you have between an extension and a device. exten => 3000,hint,SIP/olle exten => 3000,dial(SIP/olle,30) This extension in your dialplan tells Asterisk that if anyone subscribes to extension 3000, they want to know the status of SIP/olle. Without the hint, the extension will always be available and there's no notification at all. In CVS head, you can do this with IAX2 as well. I have a patch that has been in the bugtracker for a few months that adds a bit more. If you apply a call limit with the incominglimit, you will see that the notify function will tell you not only if the device is available or not, but also if they're on a call. This works beautifully with Xten's Eyebeam. I am close to securing funding for development of another addition to chan_sip during the summer. This will be the "shared line apperances" solution developed by Broadsoft and now implemented in phones from many manufacturers like Sipura, Aastra and I also suspect Grandstream and Polycom. When we have this working, you will be able to get Asterisk with one of these SIP phones to behave much more like a key system. A lot of text that really did not answer the question, sorry. /Olle Astricon - the Asterisk User's conference - Madrid June 15-17 http://www.astricon.net/europe/ - Register today! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 extension lights
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom remember always to search the wiki it has tons of info... I'm using a 360 and the lights work fine. Pay attention to the hint stuff in the dialplan. bye, M. - Original Message - From: Ross Kevlin To: asterisk-users@lists.digium.com Sent: Friday, June 03, 2005 12:35 PM Subject: [Asterisk-Users] SNOM 360 extension lights I contacted SNOM and they told me to change a couple of options but still no lights, here is what they told me Line page SIP tab:o Long SIP-Contact (RFC3840) to "off"o Support broken Registrar to "on"Advanced page:o Filter Packets from Registrar to "off" And please ask the Asterisk community for help, I'm sure they solved thatissue 100%, and we are not knowing so much about Asterisk.Your snom support Teamhas anyone gotten a 360 to work with the lights? what options and modifications to .conf files did you have to make? here are the subscribe and notifies. it seems it terminates the subscription as soon as its created. I don't think its a proxy authentication problem because it eventually sends the proxy authentication information Using latest SUBSCRIBE request as basis requestSending to 192.168.2.230 : 2051 (non-NAT)Found peer '83'Transmitting (no NAT) to 192.168.2.230:2051:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049nFrom: ;tag=z6kvtd67buTo: ;tag=as6c1cb2a5Call-ID: [EMAIL PROTECTED]CSeq: 1 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 mssip1*CLI><-- SIP read from 192.168.2.230:2051:SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rportFrom: ;tag=z6kvtd67buTo: Call-ID: [EMAIL PROTECTED]CSeq: 1 SUBSCRIBEMax-Forwards: 70Contact: Event: dialogAccept: application/dialog-info+xmlExpires: 3600Content-Length: 0 --- (12 headers 0 lines)---Ignoring this SUBSCRIBE requestFound peer '83'Transmitting (no NAT) to 192.168.2.230:2051:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049nFrom: ;tag=z6kvtd67buTo: ;tag=as6c1cb2a5Call-ID: [EMAIL PROTECTED]CSeq: 1 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 mssip1*CLI><-- SIP read from 192.168.2.230:2051:SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rportFrom: ;tag=z6kvtd67buTo: Call-ID: [EMAIL PROTECTED]CSeq: 2 SUBSCRIBEMax-Forwards: 70Contact: Event: dialogAccept: application/dialog-info+xmlProxy-Authorization: Digest username="83",realm="asterisk",nonce="16747f76",uri="sip:[EMAIL PROTECTED];user=phone",response="15d72104244317e2c0afa3499220e4ab",algorithm=md5Expires: 3600Content-Length: 0 --- (13 headers 0 lines)---Using latest SUBSCRIBE request as basis requestSending to 192.168.2.230 : 2051 (non-NAT)Found peer '83'Looking for 117 in localusers-C2021-1Transmitting (no NAT) to 192.168.2.230:2051:SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7xFrom: ;tag=z6kvtd67buTo: ;tag=as77c7b911Call-ID: [EMAIL PROTECTED]CSeq: 2 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYExpires: 3600Contact: ;expires=3600Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 361 msReliably Transmitting (no NAT) to 192.168.2.230:2051:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rportFrom: ;tag=as77c7b911To: ;tag=z6kvtd67buContact: Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: MVC 001Event: dialogContent-Type: application/dialog-info+xmlContent-Length: 203 entity="sip:[EMAIL PROTECTED]">terminated ---sip1*CLI><-- SIP read from 192.168.2.230:2051:SIP/2.0 200 OkVia: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060From: ;tag=as77c7b911To: ;tag=z6kvtd67buCall-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYContent-Length: 0 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/list
Re: [Asterisk-Users] SNOM 360 extension lights
Sorry Ross I must have missed your first postings, but what are you trying to achive? David On 03/06/05, Ross Kevlin <[EMAIL PROTECTED]> wrote: > > I contacted SNOM and they told me to change a couple of options but still no > lights, here is what they told me > > Line page SIP tab: > > o Long SIP-Contact (RFC3840) to "off" > o Support broken Registrar to "on" > > Advanced page: > > o Filter Packets from Registrar to "off" > > And please ask the Asterisk community for help, I'm sure they solved that > issue 100%, and we are not knowing so much about Asterisk. > > Your snom support Team > > has anyone gotten a 360 to work with the lights? what options and > modifications to .conf files did you have to make? > > here are the subscribe and notifies. > it seems it terminates the subscription as soon as its created. I don't > think its a proxy authentication problem > because it eventually sends the proxy authentication information > > Using latest SUBSCRIBE request as basis request > Sending to 192.168.2.230 : 2051 (non-NAT) > Found peer '83' > Transmitting (no NAT) to 192.168.2.230:2051: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n > From: ;tag=z6kvtd67bu > To: ;tag=as6c1cb2a5 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 SUBSCRIBE > User-Agent: MVC 001 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76" > Content-Length: 0 > > > --- > Scheduling destruction of call > '[EMAIL PROTECTED]' in 15000 ms > sip1*CLI> > <-- SIP read from 192.168.2.230:2051: > SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP > 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rport > From: ;tag=z6kvtd67bu > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 1 SUBSCRIBE > Max-Forwards: 70 > Contact: > Event: dialog > Accept: application/dialog-info+xml > Expires: 3600 > Content-Length: 0 > > > --- (12 headers 0 lines)--- > Ignoring this SUBSCRIBE request > Found peer '83' > Transmitting (no NAT) to 192.168.2.230:2051: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n > From: ;tag=z6kvtd67bu > To: ;tag=as6c1cb2a5 > Call-ID: [EMAIL PROTECTED] > CSeq: 1 SUBSCRIBE > User-Agent: MVC 001 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: > Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76" > Content-Length: 0 > > > --- > Scheduling destruction of call > '[EMAIL PROTECTED]' in 15000 ms > sip1*CLI> > <-- SIP read from 192.168.2.230:2051: > SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP > 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rport > From: ;tag=z6kvtd67bu > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 2 SUBSCRIBE > Max-Forwards: 70 > Contact: > Event: dialog > Accept: application/dialog-info+xml > Proxy-Authorization: Digest > username="83",realm="asterisk",nonce="16747f76",uri= > "sip:[EMAIL > PROTECTED];user=phone",response="15d72104244317e2c0afa3499220e4ab",a > lgorithm=md5 > Expires: 3600 > Content-Length: 0 > > > --- (13 headers 0 lines)--- > Using latest SUBSCRIBE request as basis request > Sending to 192.168.2.230 : 2051 (non-NAT) > Found peer '83' > Looking for 117 in localusers-C2021-1 > Transmitting (no NAT) to 192.168.2.230:2051: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x > From: ;tag=z6kvtd67bu > To: ;tag=as77c7b911 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 SUBSCRIBE > User-Agent: MVC 001 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Expires: 3600 > Contact: ;expires=3600 > Content-Length: 0 > > > --- > Scheduling destruction of call > '[EMAIL PROTECTED]' in 361 ms > Reliably Transmitting (no NAT) to 192.168.2.230:2051: > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport > From: ;tag=as77c7b911 > To: ;tag=z6kvtd67bu > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > User-Agent: MVC 001 > Event: dialog > Content-Type: application/dialog-info+xml > Content-Length: 203 > > > version="0" state="full" > entity="sip:[EMAIL PROTECTED]"> > > terminated > > > > --- > sip1*CLI> > <-- SIP read from 192.168.2.230:2051: > SIP/2.0 200 Ok > Via: SIP/2.0/UDP > 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060 > From: ;tag=as77c7b911 > To: ;tag=z6kvtd67bu > Call-ID: [EMAIL PROTECTED] > CSeq: 102 NOTIFY > Content-Length: 0 > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visi
Re: [Asterisk-Users] SNOM 360 extension lights
Ross Kevlin wrote: > I have a snom 360 that im trying to get the extension lights working i can > see the subscription being sent and a reply but the reply is a terminate. Terminate being? /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 extension lights
Ross Kevlin wrote: > the subscription is sent a reply and the reply has content that indicates > its state is terminated > > From: ;tag=as77c7b911 > To: ;tag=z6kvtd67bu > Contact: > Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360 > CSeq: 102 NOTIFY > User-Agent: MVC 001 > Event: dialog > Content-Type: application/dialog-info+xml > Content-Length: 203 > > >version="0" state="full" > entity="sip:83 at 192.168.2.252"> > > terminated > > > > from what I understand the terminated state is the end of dialog and the > subscription should end, but I still see the subscription in asterisk. > > >>Ross Kevlin wrote: >> I have a snom 360 that i'm trying to get the extension lights working I > > can > >>see the subscription being sent and a reply but the reply is a terminate. >>Terminate being? > Please show us a full packet and tell us who sends it where? Asterisk currently does only SEND notifications, we do not parse incoming notifications. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 NOTIFY syntax
On Mon, Jul 11, 2005 at 01:16:08PM -0500, Patrick Friedel wrote: > I'm rolling out an installation with snom 360s in the near future. > Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a > snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I > have the 360's set up to subscribe and notify for the line use lights, > which works like a charm for interoffice calling (between the 360's, > anyway. The IAXy, 200 and, softphone will be used by less phone > dependant types) but what I can't figure out from the Wiki is if it's > possible to have the ZAP lines notify for the outbound lines so we can > see how many lines are in use. I am by no means an expert at this, but I did some experimentation and it appears that the NOTIFY will not get sent for the trunk lines, only for extensions. I also found that the SUBSCRIBE/NOTIFY sequence only works for SIP and ZAP, I couldn't get it to work for IAX2. I do not know why this would be, and it is possible I was doing something wrong, but for what it's worth, that's my experience so far. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 and parking
On Tuesday 12 July 2005 23:29, Patrick Friedel wrote:
> OK, last showstopper that I just can't puzzle my way through - parking
> calls with the snom phones. I get the two phones connected, I hit
> transfer on one, the other phone goes to MOH and the first phone gives
> me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM
> hangs up before I have a chance to hear which extension it parked to.
> Is there a way to make the SNOM phones stay off hook until you
> explicitly hang up during a transfer? (my only complaint about these
> phones - occasionally they're just too darn smart for their own good.)
>
> I can live without actual snom-style orbits at this time (handy though
> they might be), since the current system involves parking the call on an
> external line and walking over to another office to say that they have a
> call. I imagine that down the road it'll usually just be an attended
> transfer, but we do park calls around phones a fair bit as we brainstorm
> issues.
>
> (Actually, I can't get attended transfers working, either. All
> transfers are blind. Related?)
Press hold (R on the old phones) or the line key (the buttons on the right) of
the line which you are currently using. The first call is on hold. Now call
the transfer target. Talk to him. Now press the transfer button or Xfer soft
button (below the screen). Depending on the configuration of your phone the
two call will be joined immediately ('Call join on Xfer' on the advanced), or
you have to select the target for transfer manually and press OK one more
time (<- good protection against joining your girlfriend with your wife ;) ).
This is how attended transfer works the "snom way".
Regards
Nils
--
snom technology AGGradestr. 46D-12347 Berlin
Nils Ohlmeier
mailto:[EMAIL PROTECTED] http://www.snom.com
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Re: [Asterisk-Users] SNOM 360 and parking
On 7/12/05, Patrick Friedel <[EMAIL PROTECTED]> wrote: > OK, last showstopper that I just can't puzzle my way through - parking > calls with the snom phones. I get the two phones connected, I hit > transfer on one, the other phone goes to MOH and the first phone gives > me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM > hangs up before I have a chance to hear which extension it parked to. > Is there a way to make the SNOM phones stay off hook until you > explicitly hang up during a transfer? (my only complaint about these > phones - occasionally they're just too darn smart for their own good.) The 'Transfer' button on snom phones is for blind transfer. What you describe above is the blind transfer of the caller to extension 700. > I can live without actual snom-style orbits at this time (handy though > they might be), since the current system involves parking the call on an > external line and walking over to another office to say that they have a > call. I imagine that down the road it'll usually just be an attended > transfer, but we do park calls around phones a fair bit as we brainstorm > issues. I assume from the lack of any other detail that "700" is how you park your calls? > (Actually, I can't get attended transfers working, either. All > transfers are blind. Related?) For attended transfer on the snom, use the 'hold' button instead of the 'transfer' button. IIRC the default behaviour is to transfer on hangup as you describe. We do not use call parking at-all here - it only confused the "normal" users. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 and parking
To do attended transfers with Snom 360, you need to put the current call on hold, dial the dest extension, tell him/her something, and press the Transfer button. I don't think it'll work with asterisk call parking, though... Julian J. M. On 7/12/05, Patrick Friedel <[EMAIL PROTECTED]> wrote: > OK, last showstopper that I just can't puzzle my way through - parking > calls with the snom phones. I get the two phones connected, I hit > transfer on one, the other phone goes to MOH and the first phone gives > me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM > hangs up before I have a chance to hear which extension it parked to. > Is there a way to make the SNOM phones stay off hook until you > explicitly hang up during a transfer? (my only complaint about these > phones - occasionally they're just too darn smart for their own good.) > > I can live without actual snom-style orbits at this time (handy though > they might be), since the current system involves parking the call on an > external line and walking over to another office to say that they have a > call. I imagine that down the road it'll usually just be an attended > transfer, but we do park calls around phones a fair bit as we brainstorm > issues. > > (Actually, I can't get attended transfers working, either. All > transfers are blind. Related?) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 360 audio garbled
Michael George wrote: > I have a new snom 360 on an internal net to my * box. When putting a call on > hold and taking it off, the audio will usually be broken and not > understandable. > > Sometimes this happens on incoming calls and almost always on outgoing calls. > > Anyone run into this before? > > Thx! You might try checking which codec is in use pre-hold and post-hold. On our Snom 190s, g726 always seems to sound garbled, and the call may be starting with one codec (like ulaw), then continuing with another (like g726) after being taken off of hold. Scott ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 360 audio garbled
On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote: > > You might try checking which codec is in use pre-hold and post-hold. On > our Snom 190s, g726 always seems to sound garbled, and the call may be > starting with one codec (like ulaw), then continuing with another (like > g726) after being taken off of hold. Do you mean which codec asterisk thinks it's using or which codec the phone is using at a particular time? I know that "sip show channels" will list the codecs used on a channel by *, but how do I find out what codec the Snom is using on an ongoing call? Also, if the snom went to a different codec, wouldn't the audio be completely incomprehensible? What we get is garbled and we can hear what sound like syllables, but you cannot really make it out. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 360 audio garbled
Michael George wrote: > On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote: >>You might try checking which codec is in use pre-hold and post-hold. On >>our Snom 190s, g726 always seems to sound garbled, and the call may be >>starting with one codec (like ulaw), then continuing with another (like >>g726) after being taken off of hold. > > > Do you mean which codec asterisk thinks it's using or which codec the phone is > using at a particular time? > > I know that "sip show channels" will list the codecs used on a channel by *, > but how do I find out what codec the Snom is using on an ongoing call? > > Also, if the snom went to a different codec, wouldn't the audio be completely > incomprehensible? What we get is garbled and we can hear what sound like > syllables, but you cannot really make it out. > > Thank you. Yes, I mean running "sip show channels" both pre- and post-hold and seeing if the codec listed for the channel is different. If it is, try disallowing the second codec in sip.conf. Scott ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 360 audio garbled
On Tue, Jul 19, 2005 at 02:17:07PM -0400, Scott wrote: > > Do you mean which codec asterisk thinks it's using or which codec the phone > > is > > using at a particular time? > > > > I know that "sip show channels" will list the codecs used on a channel by *, > > but how do I find out what codec the Snom is using on an ongoing call? > > > > Also, if the snom went to a different codec, wouldn't the audio be > > completely > > incomprehensible? What we get is garbled and we can hear what sound like > > syllables, but you cannot really make it out. > > > > Thank you. > > Yes, I mean running "sip show channels" both pre- and post-hold and > seeing if the codec listed for the channel is different. If it is, try > disallowing the second codec in sip.conf. Okay, I'll check that. Is there a way to disable all but the first codec from the GUI? I've not yet gotten to the mass-configuration yet... Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 360 audio garbled
On Tue, Jul 19, 2005 at 02:17:07PM -0400, Scott wrote: > Michael George wrote: > > On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote: > >>You might try checking which codec is in use pre-hold and post-hold. On > >>our Snom 190s, g726 always seems to sound garbled, and the call may be > >>starting with one codec (like ulaw), then continuing with another (like > >>g726) after being taken off of hold. > > > > > > Do you mean which codec asterisk thinks it's using or which codec the phone > > is > > using at a particular time? > > > > I know that "sip show channels" will list the codecs used on a channel by *, > > but how do I find out what codec the Snom is using on an ongoing call? > > > > Also, if the snom went to a different codec, wouldn't the audio be > > completely > > incomprehensible? What we get is garbled and we can hear what sound like > > syllables, but you cannot really make it out. > > > > Thank you. > > Yes, I mean running "sip show channels" both pre- and post-hold and > seeing if the codec listed for the channel is different. If it is, try > disallowing the second codec in sip.conf. According to the distributor, the lastest firmware revisions addressed this problem. I have now upgraded the firmware and I hope it helps. I'll see later this week. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
Patrick Friedel wrote: I assume the first one is the snom requesting the PBX to start recording, and the second is asterisk reminding the snom that it doesn't allow the INFO method and to get bent? Yep, you're right. That would be an interesting addition to Asterisk, but it will likely require a bounty to get someone to write support for it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
Patrick Friedel wrote: > 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 mention I _see_ on the wiki is "one touch RECORD > button usuable only with special PBX support via SIP INFO method" which > isn't much of an answer. > Is this during a call? Can you please send me a full SIP DEBUG of the call? Brainstorming, maybe we could treat this as a transfer to a local extension somehow and turn monitor on in the dial plan that way... /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
Olle E. Johansson wrote: Patrick Friedel wrote: 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 mention I _see_ on the wiki is "one touch RECORD button usuable only with special PBX support via SIP INFO method" which isn't much of an answer. Is this during a call? Can you please send me a full SIP DEBUG of the call? Brainstorming, maybe we could treat this as a transfer to a local extension somehow and turn monitor on in the dial plan that way... Yeah, that was in the middle of a call - the only other SIP debug information is the normal call build up and tear down. I can generate it if you want, but it's nothing exciting, just the usual handshaking. But that was kind of what I was thinking would be a solution - Asterisk sees the Record INFO packet, and conferences the call to a local extension with Monitor() going. I'm not 100% sure whether or not the Snom 360 expects anything _else_ (other than a simple acknowledgement) back from the PBX, as there doesn't appear to be a whitepaper for it it on Snom's site. I would assume it would be fairly straightforward in chan_sip.c to bang in a new method, but whether there are any ramifications, well... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
> Is this during a call? Can you please send me a full SIP DEBUG of the call? > > Brainstorming, maybe we could treat this as a transfer to a local > extension somehow and turn monitor on in the dial plan that way... Hmm IMO the automon-feature would be better for this. It does exactly what we want (start recording during a call) and is configurable via Dial-Options. The only thing I don't know is how to activate it without sending the DTMF sequence. Maik Schmitt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 360 record button?
It would be nice if the PBX can acknowlegdge the Record header - then it would have the chance to paint a record icon on the screen. In the next release.-) CS > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Patrick Friedel > Sent: Thursday, July 28, 2005 7:31 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Snom 360 record button? > > Olle E. Johansson wrote: > > >Patrick Friedel wrote: > > > > > >>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 mention I _see_ on the wiki is > "one touch > >>RECORD button usuable only with special PBX support via SIP INFO > >>method" which isn't much of an answer. > >> > >> > >> > >Is this during a call? Can you please send me a full SIP > DEBUG of the call? > > > >Brainstorming, maybe we could treat this as a transfer to a local > >extension somehow and turn monitor on in the dial plan that way... > > > > > Yeah, that was in the middle of a call - the only other SIP > debug information is the normal call build up and tear down. > I can generate it if you want, but it's nothing exciting, > just the usual handshaking. > But that was kind of what I was thinking would be a solution > - Asterisk sees the Record INFO packet, and conferences the > call to a local extension with Monitor() going. I'm not 100% > sure whether or not the Snom 360 expects anything _else_ > (other than a simple acknowledgement) back from the PBX, as > there doesn't appear to be a whitepaper for it it on Snom's > site. I would assume it would be fairly straightforward in > chan_sip.c to bang in a new method, but whether there are any > ramifications, well... > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
Maik Schmitt wrote: Is this during a call? Can you please send me a full SIP DEBUG of the call? Brainstorming, maybe we could treat this as a transfer to a local extension somehow and turn monitor on in the dial plan that way... Hmm IMO the automon-feature would be better for this. It does exactly what we want (start recording during a call) and is configurable via Dial-Options. The only thing I don't know is how to activate it without sending the DTMF sequence. Hmm, you're right, I wasn't aware of the automon feature - I don't know if it was in the original SIP trace, but the 360 is sending _some_ DTMF signalling, but I don't know what it's actually sending. I'm currently at 9 for verbosity, let me amp that up and see if it will ever actually display the tones it's receiving. voip*CLI> sip debug peer pjf SIP Debugging Enabled for IP: 10.0.1.213:2051 voip*CLI> set verbose 255 Verbosity was 9 and is now 255 voip*CLI> [boring build up that isn't new to anyone] Sip read: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih;rport From: "Patrick" ;tag=4rbnk4yyxd To: ;tag=as3b19835c Call-ID: [EMAIL PROTECTED] CSeq: 3 INFO Max-Forwards: 70 Contact: User-Agent: snom360/3.60r Record: on Content-Length: 0 11 headers, 0 lines Receiving DTMF! Jul 28 14:12:15 WARNING[26025]: chan_sip.c:6166 receive_info: Unable to parse INFO message from [EMAIL PROTECTED] Content Transmitting (no NAT): SIP/2.0 415 Unsupported media type Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih From: "Patrick" ;tag=4rbnk4yyxd To: ;tag=as3b19835c Call-ID: [EMAIL PROTECTED] CSeq: 3 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 [teardown] Hmm. It's getting _something_, but the verbosity won't reveal it. Checking chan_sip.c, I don't see a mechanism for revealing the data, it'd be around line 7716. It we could reveal it (or Nils just tells us. :), it might be as simple as editing features.conf and setting automon to whatever DTMF the Snom sends when you hit the record button. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
On Thursday 28 July 2005 17:12, Patrick Friedel wrote: > Yeah, that was in the middle of a call - the only other SIP debug > information is the normal call build up and tear down. I can generate > it if you want, but it's nothing exciting, just the usual handshaking. > But that was kind of what I was thinking would be a solution - Asterisk > sees the Record INFO packet, and conferences the call to a local > extension with Monitor() going. I'm not 100% sure whether or not the > Snom 360 expects anything _else_ (other than a simple acknowledgement) > back from the PBX, as there doesn't appear to be a whitepaper for it it > on Snom's site. I would assume it would be fairly straightforward in Sorry for the missing whitepaper ;-) The snom phone is not expecting anything else then any reply for the INFO request. Nils > chan_sip.c to bang in a new method, but whether there are any > ramifications, well... -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
Nils Ohlmeier wrote: On Thursday 28 July 2005 17:12, Patrick Friedel wrote: Yeah, that was in the middle of a call - the only other SIP debug information is the normal call build up and tear down. I can generate it if you want, but it's nothing exciting, just the usual handshaking. But that was kind of what I was thinking would be a solution - Asterisk sees the Record INFO packet, and conferences the call to a local extension with Monitor() going. I'm not 100% sure whether or not the Snom 360 expects anything _else_ (other than a simple acknowledgement) back from the PBX, as there doesn't appear to be a whitepaper for it it on Snom's site. I would assume it would be fairly straightforward in Sorry for the missing whitepaper ;-) The snom phone is not expecting anything else then any reply for the INFO request. Heh, figured that would draw you out of the woodwork. It's good to know that information. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
Christian Stredicke wrote: > It would be nice if the PBX can acknowlegdge the Record header - then it > would have the chance to paint a record icon on the screen. > > In the next release.-) > Right. Is there another header for turning off recording? Anyway, we should not send "unsupported media type"... /O ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 360 locked up
Try loading
http://phone-ip-address/line_sip.htm?settings=save&user_dp_str1= (if
that was in the line 1) while the phone boots up (keep your finger on
the reload button). If that does not work, you need to do a tftp update.
Also consider moving to version 4.5
(http://www.snom.com/snom360_release_notes.html).
Christian
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steven Ringwald
> Sent: Friday, December 23, 2005 5:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] SNOM 360 locked up
>
> Hello all!
>
> I was trying to get the dial-string setup for my regular
> usage, and the phone locked up in the middle of dialing.
> Basically, I put the following line in, hit save, and got as
> far as dialing '9', and the phone froze.
>
> |^(9[0-9]{10}|sip:[EMAIL PROTECTED]|d
>
> Now the phone boots up to the SNOM splash screen and hangs
> there. I can ping it, but cannot get to the web-interface and
> cannot reset to factory defaults using the web-gui.
>
> Any idea how I can reset the phone to factory w/o using the
> GUI? Or am I completely hosed?
>
> Steve
>
>
>
>
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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RE: [Asterisk-Users] SNOM 360 locked up
On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote: Try loading http://phone-ip-address/line_sip.htm?settings=save&user_dp_str1= (if that was in the line 1) while the phone boots up (keep your finger on the reload button). If that does not work, you need to do a tftp update. Yeah. The website address didn't work. (The phone, I think, is not far enough along to even start the webserver). I will try the tftp update method, and see what happens. So far, though, it doesn't seem to be hitting the tftp server that I set up manually. Also consider moving to version 4.5 (http://www.snom.com/snom360_release_notes.html). Any idea how to do that? I think it is running 4.1. I have put the firmware image URL into the upgrade line before, and it didn't take. (Ended up going back to what it had previously had). Thanks for the help! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 locked up
On Friday 23 December 2005 00:39, Steven Ringwald wrote: > On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote: > > Try loading > > http://phone-ip-address/line_sip.htm?settings=save&user_dp_str1= (if > > that was in the line 1) while the phone boots up (keep your finger on > > the reload button). If that does not work, you need to do a tftp update. > > Yeah. The website address didn't work. (The phone, I think, is not far > enough along to even start the webserver). I will try the tftp update > method, and see what happens. > > So far, though, it doesn't seem to be hitting the tftp server that I set > up manually. A step by step description can be found here: http://www.snom.com/wiki/index.php/Main_Page#Firmware_Update > > > Also consider moving to version 4.5 > > (http://www.snom.com/snom360_release_notes.html). > > Any idea how to do that? I think it is running 4.1. I have put the > firmware image URL into the upgrade line before, and it didn't take. > (Ended up going back to what it had previously had). > > Thanks for the help! > Steve Regards, Sven ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
To add to this - when I do a 'show hints' the phone shows state 0, which I assume means not in use. PaulH On Thu, 2005-09-01 at 12:38 +1000, Paul Hales wrote: > I am setting up a snom 360, and the lights come on OK when the mapped > user makes an outgoing call, but when the user takes an incoming call > the light does not come on. > > I do not want to install the bristuff patch if possible. > (although I can see that with the devstate command I can make the lights > do whatever I want) > > Any ideas? > > regards, > > Paul Hales > Melbourne, Australia > > CAUTION: This email message and accompanying data may contain information > that is confidential. If you are not the intended recipient, you are notified > that any use, dissemination, distribution or copying of this message or data > is prohibited. If you have received this email message in error, please > notify us immediately and erase all copies of this message and attachments. > Thank you. > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > CAUTION: This email message and accompanying data may contain information > that is confidential. If you are not the intended recipient, you are notified > that any use, dissemination, distribution or copying of this message or data > is prohibited. If you have received this email message in error, please > notify us immediately and erase all copies of this message and attachments. > Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
Hello Paul, Thursday, September 1, 2005, 4:38:42 AM, you wrote: 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) Same here, it think it depends on hint status: when you make a call calling hint is set to 1, but called one stays 0. Correct behaviour should be put the hint of the caller to 1 (steady ligt) while calling put the hint of the called to X (blinking light, cant remember which state it is ) while phone is ringing, then to 1 if call is answered. Unfortunately I dont know how to accomplish this Regards! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
Issue #3644 has recently been committed to CVS-HEAD which allows for full device state notification via subscriptions for Snom 360 and other supporting phones w/o the need for additional patches. On 9/1/05, Alessio Focardi <[EMAIL PROTECTED]> wrote: Hello Paul,Thursday, September 1, 2005, 4:38:42 AM, you wrote: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 callPH> 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)Same here, it think it depends on hint status: when you make a callcalling hint is set to 1, but called one stays 0.Correct behaviour should beput the hint of the caller to 1 (steady ligt) while calling put the hint of the called to X (blinking light, cant remember whichstate it is ) while phone is ringing, then to 1 if call is answered.Unfortunately I dont know how to accomplish this Regards! --Best regards,Alessiomailto:[EMAIL PROTECTED]___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
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 "Filter Packets from Registrar" turned off (under Advanced). Next, make sure you have hint priorities setup for each of the extensions you are trying to monitor. With both of these in place, you should see an entry for each extension you are monitoring when you do "sip show subscriptions" from the * CLI. If not, rinse and repeat the above steps. Also, you may want to manually recycle power on the 360 if you happen to reset asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to wait until the phones resend the subscription. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
IS there a way to make the phone reboot each day at a time? On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote: > 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 "Filter Packets from Registrar" turned off > (under Advanced). Next, make sure you have hint priorities setup for each of > the extensions you are trying to monitor. With both of these in place, you > should see an entry for each extension you are monitoring when you do "sip > show subscriptions" from the * CLI. If not, rinse and repeat the above > steps. Also, you may want to manually recycle power on the 360 if you happen > to reset asterisk for any reason (reload extensions, etc), as it will lose > all the subscriptions and have to wait until the phones resend the > subscription. > > -Jeff > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
Yes. You could send a sip notify via asterisk and set that up via cron to happen once per day. eg. "asterisk -rx "sip notify reboot-snom " Make sure that the reboot-snom clause is setup in sip_notify.conf before attempting this. On 9/1/05, altus <[EMAIL PROTECTED]> wrote: IS there a way to make the phone reboot each day at a time?On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote: > 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 "Filter Packets from Registrar" turned off (under Advanced). Next, make sure you have hint priorities setup for each of the extensions you are trying to monitor. With both of these in place, you should see an entry for each extension you are monitoring when you do "sip show subscriptions" from the * CLI. If not, rinse and repeat the above steps. Also, you may want to manually recycle power on the 360 if you happen to reset asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to wait until the phones resend the subscription. >> -Jeff>> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list > Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>--Thanks Altus SnymanStormcorp Network Solutions+27 11 8071141 exten 301___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
If you do not want to buy a timer based power supply ;) you can send a NOTIFY with 'Event: reboot' to the phone. Nils On Thursday 01 September 2005 14:41, altus wrote: > IS there a way to make the phone reboot each day at a time? > > On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote: > > 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 "Filter Packets from Registrar" turned off > > (under Advanced). Next, make sure you have hint priorities setup for > > each of the extensions you are trying to monitor. With both of these in > > place, you should see an entry for each extension you are monitoring when > > you do "sip show subscriptions" from the * CLI. If not, rinse and repeat > > the above steps. Also, you may want to manually recycle power on the 360 > > if you happen to reset asterisk for any reason (reload extensions, etc), > > as it will lose all the subscriptions and have to wait until the phones > > resend the subscription. > > > > -Jeff > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
IS there a way to make the phone reboot each day at a time? You could do it via a cron job by wget'ting the reboot uri (on the advanced page again), but there really shouldn't be any need to do so. The only time subscriptions should "disappear" is when you do a reload or restart on asterisk. Even after a reload or restart the subscriptions will come back, but it usually takes ~30 minutes or so depending on when the last subscriptions were sent. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
SO if I do a reboot of the system each night at 12,it should be up and working again at 8 in the morning? On Thu, 2005-09-01 at 08:49 -0500, Jeff Brownlee wrote: > >> IS there a way to make the phone reboot each day at a time? > > You could do it via a cron job by wget'ting the reboot uri (on the advanced > page again), > but there really shouldn't be any need to do so. The only time subscriptions > should > "disappear" is when you do a reload or restart on asterisk. Even after a > reload or restart > the subscriptions will come back, but it usually takes ~30 minutes or so > depending > on when the last subscriptions were sent. > > -Jeff > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
My experience with auto-rebooting schemes is that reliability doesn't improve. I also reported the non-registration of the Snom phones as a bug. On a few occasions I found that the phone lost registration, and rebooting or power cycling the phone didn't help (athorization failed at the * console) but re-entering the password in the phone did. Would this match your problem too? I guess it would be nice if we could make * log each password with which a SIP client tries to register (yes I am aware of the security implications of it). My guess is that the password disappears from the phone or it is corrupted. On Fri, 2 Sep 2005, altus wrote: SO if I do a reboot of the system each night at 12,it should be up and working again at 8 in the morning? On Thu, 2005-09-01 at 08:49 -0500, Jeff Brownlee wrote: IS there a way to make the phone reboot each day at a time? You could do it via a cron job by wget'ting the reboot uri (on the advanced page again), but there really shouldn't be any need to do so. The only time subscriptions should "disappear" is when you do a reload or restart on asterisk. Even after a reload or restart the subscriptions will come back, but it usually takes ~30 minutes or so depending on when the last subscriptions were sent. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
We schedule a reboot each night at 11/12 so it clears any errors,or hanging channels,in case for save keeping Im just not sure then with the 360 that it would keep the panel lights working. Got those problems your talking about,lost with the 220's On Fri, 2005-09-02 at 09:39 +0200, Remco Barende wrote: > My experience with auto-rebooting schemes is that reliability doesn't > improve. > > I also reported the non-registration of the Snom phones as a bug. On a few > occasions I found that the phone lost registration, and rebooting or power > cycling the phone didn't help (athorization failed at the * console) but > re-entering the password in the phone did. > > Would this match your problem too? > > I guess it would be nice if we could make * log each password with > which a SIP client tries to register (yes I am aware of the security > implications of it). My guess is that the password disappears from the > phone or it is corrupted. > > > On Fri, 2 Sep 2005, altus wrote: > > > SO if I do a reboot of the system each night at 12,it should be up and > > working again at 8 in the morning? > > > > On Thu, 2005-09-01 at 08:49 -0500, Jeff Brownlee wrote: > IS there a way to make the phone reboot each day at a time? > >> > >> You could do it via a cron job by wget'ting the reboot uri (on the > >> advanced page again), > >> but there really shouldn't be any need to do so. The only time > >> subscriptions should > >> "disappear" is when you do a reload or restart on asterisk. Even after a > >> reload or restart > >> the subscriptions will come back, but it usually takes ~30 minutes or so > >> depending > >> on when the last subscriptions were sent. > >> > >> -Jeff > >> > >> ___ > >> --Bandwidth and Colocation sponsored by Easynews.com -- > >> > >> Asterisk-Users mailing list > >> Asterisk-Users@lists.digium.com > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
On Friday 02 September 2005 10:05, altus wrote: > We schedule a reboot each night at 11/12 so it clears any errors,or > hanging channels,in case for save keeping > Im just not sure then with the 360 that it would keep the panel lights > working. That depends on the behavior of * when restarting. If * tells the phones via a NOTIFY that their subscriptions are not valid any more, the phones will try get new subscriptions. If * does not tell phones about the lost subscriptions (which I fear), then it depends on * how it reacts on the re-subscription of phones. Because the phones would still belive to have a subscription and should send sooner or later a SUBSCRIBE with existing To-tag. Now it depends on * how it reacts on such a re-newale of a non-existing (from * perspective) subscription. Regards Nils Ohlmeier -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 hold problem
Sorry, but we are unable to reproduce your problem description here locally (without an *). If you are able to reproduce it, we guess it must be related to the *. In this case it could help if you send us SIP-traces of that scenario, so that we can look into this problem. Regards Nils Ohlmeier On Thursday 01 September 2005 16:45, Michael George wrote: > Hello, > > I have a customer who said that their Snom 360 is joining calls by > accident. > > The situation is that they had one call on the line and another call came > in. She pressed the hold button on the phone and the two calls were joined > together. > > I do have "Call join on Xfer" set to yes, but I thought that would only > come into play when doing a transfer, not putting someone on hold. > > The phone is at firmware 4.1, and there are no new updates, so that > shouldn't be it. > > Anyone else experience this behavior on the phones, or know if I need to > turn off Call Join on Xfer? > > Thanks! -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 360 Passsword Issue
In the Snom web management page under Advanced make sure "Challenge response on phone" is turned to OFF. This is a stupid feature to have on by default from the factory. -Original Message- From: Edward de Zeeuw [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 21, 2006 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Snom 360 Passsword Issue I have multiple (20+) Snom 360 phones communicating with asterisk 1.2.7.1. Almost regularly (daily) and in some cases ongoing 9every 10 minutes the phones ask for password and id the account they are seeking the password for. If I hit the X key the phone continues operating normally. Has anyone else come across a similar issue? Edward ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Passsword Issue
turn challenge_response off On 6/21/06, Edward de Zeeuw <[EMAIL PROTECTED]> wrote: I have multiple (20+) Snom 360 phones communicating with asterisk 1.2.7.1. Almost regularly (daily) and in some cases ongoing 9every 10 minutes the phones ask for password and id the account they are seeking the password for. If I hit the X key the phone continues operating normally. Has anyone else come across a similar issue? Edward ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Passsword Issue
I'll take a look first thing tomorrow and let you know what I find. Thanks! Edward Colin Anderson wrote: > In the Snom web management page under Advanced make sure "Challenge response > on phone" is turned to OFF. This is a stupid feature to have on by default > from the factory. > > -Original Message- > From: Edward de Zeeuw [mailto:[EMAIL PROTECTED] > Sent: Wednesday, June 21, 2006 11:54 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Snom 360 Passsword Issue > > > I have multiple (20+) Snom 360 phones communicating with asterisk > 1.2.7.1. Almost regularly (daily) and in some cases ongoing 9every 10 > minutes the phones ask for password and id the account they are seeking > the password for. If I hit the X key the phone continues operating > normally. Has anyone else come across a similar issue? > > Edward > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Passsword Issue
I have had the same problem too, I solved resetting the phone to factory defaults Edward de Zeeuw wrote: I'll take a look first thing tomorrow and let you know what I find. Thanks! Edward Colin Anderson wrote: In the Snom web management page under Advanced make sure "Challenge response on phone" is turned to OFF. This is a stupid feature to have on by default from the factory. -Original Message- From: Edward de Zeeuw [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 21, 2006 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Snom 360 Passsword Issue I have multiple (20+) Snom 360 phones communicating with asterisk 1.2.7.1. Almost regularly (daily) and in some cases ongoing 9every 10 minutes the phones ask for password and id the account they are seeking the password for. If I hit the X key the phone continues operating normally. Has anyone else come across a similar issue? Edward ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Hinting tricks
On 3/6/06, Colin Anderson <[EMAIL PROTECTED]> wrote: > I was always puzzled by posts to the list about people having problems > getting hints to work on a Snom, since I always seem to have no problem > making it work. That is, until today when I tried to get a sidecar to work. > All I could do was get a monitored extension light to light up continuously, > regardless of state. Frustrating! Going back to my working dialplans where I > got 1 or 2 lights working fine, I saw the pattern and the difference between > working and non-working, and I realized that other people were experiencing > the same problem as I was. The trick is the *order* in which you put your > hint priorities in your dialplan. My non-working sidecar dialplan had all > the hint priorities grouped together: [snip] Another "hint" for getting hints working, although this only relates to older 1.0.x versions of Asterisk (It is already fixed in 1.2.x) is that status changes are not notified for channels where there is a hyphen '-' in the channel name, so replacing all hyphens with underscores in your sip.conf section names might prove useful :) Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Hinting tricks
On Wednesday 22 March 2006 05:26, Steve Davies wrote: > Another "hint" for getting hints working, although this only relates > to older 1.0.x versions of Asterisk (It is already fixed in 1.2.x) is > that status changes are not notified for channels where there is a > hyphen '-' in the channel name, so replacing all hyphens with > underscores in your sip.conf section names might prove useful :) That certainly seems to be a bug in my opinion. If a hyphen is allowed in the name, it better damn well be allowed ANYWHERE a name is being passed. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Hinting tricks
On 3/22/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
> On Wednesday 22 March 2006 05:26, Steve Davies wrote:
> > Another "hint" for getting hints working, although this only relates
> > to older 1.0.x versions of Asterisk (It is already fixed in 1.2.x) is
> > that status changes are not notified for channels where there is a
> > hyphen '-' in the channel name, so replacing all hyphens with
> > underscores in your sip.conf section names might prove useful :)
>
> That certainly seems to be a bug in my opinion. If a hyphen is allowed in the
> name, it better damn well be allowed ANYWHERE a name is being passed.
I guess the devs agreed, as it is fixed in version 1.2.x :) In case it
helps anyone, attached is a crude replica of the 1.2 changes as
applies to 1.0.9
Steve
--- pbx.c~ 2006-03-21 11:09:31.0 +
+++ pbx.c 2006-03-21 11:11:00.0 +
@@ -1455,7 +1455,7 @@
vsnprintf(device, sizeof(device), fmt, ap);
va_end(ap);
- rest = strchr(device, '-');
+ rest = strrchr(device, '-');
if (rest) {
*rest = 0;
}
diff -ur channel.c~ channel.c
--- channel.c~ 2006-03-21 14:18:58.0 +
+++ channel.c 2006-03-21 14:12:47.0 +
@@ -1983,7 +1983,7 @@
while (chan) {
strncpy(name, chan->name, sizeof(name)-1);
ast_mutex_unlock(&chan->lock);
- cut = strchr(name,'-');
+ cut = strrchr(name,'-');
if (cut)
*cut = 0;
if (!strcmp(name, device)) {
diff -ur channels/chan_sip.c~ channels/chan_sip.c
--- channels/chan_sip.c~ 2006-03-21 14:18:58.0 +
+++ channels/chan_sip.c 2006-03-21 12:28:32.0 +
@@ -4401,7 +4401,7 @@
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", p->name);
register_peer_exten(p, 0);
p->expire = -1;
- ast_device_state_changed("SIP/%s", p->name);
+ ast_device_state_changed("SIP/%s-", p->name);
if (p->selfdestruct) {
p->delme = 1;
prune_peers();
@@ -5013,7 +5013,7 @@
}
}
if (!res) {
- ast_device_state_changed("SIP/%s", peer->name);
+ ast_device_state_changed("SIP/%s-", peer->name);
}
if (res < 0)
transmit_response(p, "403 Forbidden", &p->initreq);
@@ -6751,7 +6751,7 @@
peer->lastms = pingtime;
peer->call = NULL;
if (statechanged) {
-ast_device_state_changed("SIP/%s", peer->name);
+ast_device_state_changed("SIP/%s-", peer->name);
if (newstate == 2) {
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Lagged\r\nTime: %d\r\n", peer->name, pingtime);
} else {
@@ -8147,7 +8147,7 @@
sip_destroy(peer->call);
peer->call = NULL;
peer->lastms = -1;
- ast_device_state_changed("SIP/%s", peer->name);
+ ast_device_state_changed("SIP/%s-", peer->name);
/* Try again quickly */
peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer);
return 0;
___
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RE: [Asterisk-Users] Snom 360 Hinting tricks
I was having trouble getting hints to work with my GXP-2000 (with the beta firmware). I am running Asterisk 1.2.5. I had hyphens in the SIP channel names and it wasn't working. I have changed them to underscores and it has worked in 1.2.5. So I would say that it is not yet fixed in Asterisk <=1.2.5 Jared -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Thursday, 23 March 2006 12:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Snom 360 Hinting tricks On 3/22/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Wednesday 22 March 2006 05:26, Steve Davies wrote: > > Another "hint" for getting hints working, although this only relates > > to older 1.0.x versions of Asterisk (It is already fixed in 1.2.x) is > > that status changes are not notified for channels where there is a > > hyphen '-' in the channel name, so replacing all hyphens with > > underscores in your sip.conf section names might prove useful :) > > That certainly seems to be a bug in my opinion. If a hyphen is allowed in the > name, it better damn well be allowed ANYWHERE a name is being passed. I guess the devs agreed, as it is fixed in version 1.2.x :) In case it helps anyone, attached is a crude replica of the 1.2 changes as applies to 1.0.9 Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Hinting tricks
On 3/23/06, Jared Davison <[EMAIL PROTECTED]> wrote: > I was having trouble getting hints to work with my GXP-2000 (with the beta > firmware). I am running Asterisk 1.2.5. I had hyphens in the SIP channel > names and it wasn't working. I have changed them to underscores and it has > worked in 1.2.5. So I would say that it is not yet fixed in Asterisk <=1.2.5 > Well, It looked fixed... Perhaps it is only fixed in trunk. Still, there is a workaround in the meantime :) Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 Function Keys
> 2) One digium quadri primary ISDN interface (TE410P) > 3) Two Rhino Channel Banks > 4) 25 Analogue Phones on every channel bank > > How I can configure function keys on my SNOM 360 for monitoring analogue > phone status? I haven't used the Rhino Channel banks yet, so I'm guessing to some degree here: I'm not exactly sure how you address each phone on the channel bank. Presumably it connects to the digium card. If so, don't you have something like ZAP/1 to dial first phone ZAP/2 to dial second etc? If so, you should be able to add hints to your dialplan for each phone and make the snom monitor those. The snom360 works rather well with hinting and allows you to call/transfer a call to the monitored phone when you press the button too. For example... in the dialplan: exten => 4101,hint,Zap/1 for the functionkey (type destination) put: Conrad. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 flickering screen
My company has had a few screens go out on us, but all of those were completely blank. I'm not sure if we just got a bad batch or what, but the Snom phones are usually a solid piece of hardware. I'd try to RMA it. Nick Hoffman wrote: Hi guys. I just bought and configured a Snom 360 and have noticed that the LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). Either way, it's very distracting. Has anyone else encountered this before? Any solutions? Cheers, -- Nick E: [EMAIL PROTECTED] P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 flickering screen
I had a problem with snom where the screen went completly blank. Snom told me there was an issue where that the cable going from the phone board to the screen would fall out. I opend the phone and sliped it back in. - Original Message - From: "Nick Hoffman" <[EMAIL PROTECTED]> To: "asterisk-users Mailing List" Sent: Tuesday, November 07, 2006 10:53 AM Subject: [asterisk-users] Snom 360 flickering screen Hi guys. I just bought and configured a Snom 360 and have noticed that the LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). Either way, it's very distracting. Has anyone else encountered this before? Any solutions? Cheers, -- Nick E: [EMAIL PROTECTED] P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 flickering screen
> - Original Message - > From: "Nick Hoffman" <[EMAIL PROTECTED]> > To: "asterisk-users Mailing List" > Sent: Tuesday, November 07, 2006 10:53 AM > Subject: [asterisk-users] Snom 360 flickering screen > > > Hi guys. I just bought and configured a Snom 360 and have noticed that > > the LCD is constantly flickering at a rate of ~10-15Hz (that's a > > guess). Either way, it's very distracting. Has anyone else encountered > > this before? Any solutions? > > > > Cheers, > > -- Nick > > E: [EMAIL PROTECTED] > > P: +61 7 5591 3588 > > F: +61 7 5591 6588 > > > > If you receive this email by mistake, please notify us and do not make > > any use of the email. We do not waive any privilege, confidentiality > > or copyright associated with it. On Wed November 8 2006 01:30, "Dovid B" <[EMAIL PROTECTED]> wrote: > I had a problem with snom where the screen went completly blank. Snom > told me there was an issue where that the cable going from the phone > board to the screen would fall out. I opend the phone and sliped it back > in. Hi Dovid, thanks for the recommendation. I opened up my 360 and looked around. Everything was connected properly, but I noticed that some of the wires connecting the two PCBs were partially crushed by one of the case's support posts. When closing the case, I made sure to move the wires out of the way of the posts. The screen's much better now. If I look at it from an extreme angle I can see a lot of flickering, but at normal angles there's almost no flickering. Thanks again! -- Nick E: [EMAIL PROTECTED] P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom 360 incorrect US indications
Snom's US tones have always been Terrible.. I have contacted them several times, they recommeded I "use another countrys tones" u, I don't think so, I don't think customers will like that too much. Plus the call waiting tone is terrible, its loud, cuts out the call, and the outside party can hear the tone! It should be very light and unobtrusive like a normal phones call waiting. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, February 19, 2006 12:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] snom 360 incorrect US indications Anyone noticed the snom 360 indications are incorrect for US zone? menu->preferences->tone scheme->usa indications.conf: [general] country=us extensions.conf: exten => ,1,Answer exten => ,n,Playtones(dial) exten => ,n,Wait(30) exten => ,1,Busy exten => ,1,Answer exten => ,n,Playtones(busy) exten => ,n,Wait(30) hit speakerphone on the snom 360. listen to the dialtone. now dial and compare to asterisk's dialtone. hit speakerphone on the snom 360. dial . now compare to the busy signal you get from . in each case, snom tone is incorrect and asterisk is correct. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight & Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 4.0 firmware issue
Colin E. McDonald wrote: The new update seems to have cured my issue with calls intersecting and Zap lines not being hung up after the user terminates the session but now I am having sound issues with all of my phones. The sounds seems to be very low on all of them and there is a definite change from the same set when it was at 3.6j. The speaker also generates what appears to be static but you can discern a scratchy sounding echo. This is also occuring on all phones after the upgrade. I have genereated a support ticket to Snom but I wanted to see if anyone on the list has run into the same behavior. Thanks Colin I have about 15 snom 360 phones loaded with 4.0 - and mine seem to be working great. I did update the memory manager as well, not sure if it helped with the issues you mention because I loaded it right after 4.0: http://snom.com/download/share/snom360-3.31-r.bin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 360 locked up SOLVED
Thank you so much for your help, Christian! Your suggestion worked perfectly, and the phones came back up without a problem. Froehe Weihnachten! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 360 locked up SOLVED
He had run into a deadlock situation where he entered an (illegal) string for the dial plan that made the phone lock up right after reboot. That bug was fixed in one of the early 4.x versions. The way out was a little trick with the web browser. Generally I think if people have a problem today they should move to 4.5. This version seems to be pretty stable, we did not get any crash-complains or major problem reports from this version. For those who want to move on (feature-wise), it is time to jump on the 5.x train - the 5.0 version has been released a few days ago. We tried our best to test this version as good as possible (including an Asterisk-lab test), but from experience we know that new features always take a certain time to stabilise. Therefore, I would today move to 5.0 only if it has a feature that the 4.5 does not have. We tried to keep the release notes as informative as possible to make this decision as easy as possible for you. Happy New Year! Christian > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Michael George > Sent: Thursday, December 29, 2005 10:17 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SNOM 360 locked up SOLVED > > On Thu, Dec 22, 2005 at 03:58:07PM -0800, Steven Ringwald wrote: > > Thank you so much for your help, Christian! Your suggestion worked > > perfectly, and the phones came back up without a problem. > > What part of his suggestion? Upgrading the firmware to 4.5 > via the tftp server? > > Please elaborate for the benefit of others who may run into > this problem. > > Thank you. > > -- > -M > > There are 10 kinds of people in this world: > Those who can count in binary and those who cannot. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 locked up SOLVED
Christian Stredicke writes: > Generally I think if people have a problem today they should move to > 4.5. This version seems to be pretty stable, we did not get any > crash-complains or major problem reports from this version. > > For those who want to move on (feature-wise), it is time to jump on the > 5.x train - the 5.0 version has been released a few days ago. We tried > our best to test this version as good as possible (including an > Asterisk-lab test), but from experience we know that new features always > take a certain time to stabilise. Therefore, I would today move to 5.0 > only if it has a feature that the 4.5 does not have. Upgraded to 5.0 earlier this week and love it. Biggest reason for the move here was that our Snom 320 was dropping calls from time to time, sometimes immediately on pickup, other times mid conversation. We have not seen that since going to 5.0 earlier this week. Janina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 locked up SOLVED
On Thu, Dec 22, 2005 at 03:58:07PM -0800, Steven Ringwald wrote: > Thank you so much for your help, Christian! Your suggestion worked > perfectly, and the phones came back up without a problem. What part of his suggestion? Upgrading the firmware to 4.5 via the tftp server? Please elaborate for the benefit of others who may run into this problem. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 360 with Firmware 6.1?
Just installed! Use 6.1.1 (beta) because 6.1 has a few of registration problems. Bye > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Koopmann, Jan-Peter > Sent: Friday, June 23, 2006 9:24 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Snom 360 with Firmware 6.1? > > Hi, > > Has anybody experience with Snom360 and Firmware 6.X with > Asterisk 1.2.X? I am currently using Firmware 5.5 without > serious problems but wanted to make sure 6.X will work as > well (including subscription etc.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 with Firmware 6.1?
Koopmann, Jan-Peter wrote: Hi, Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I am currently using Firmware 5.5 without serious problems but wanted to make sure 6.X will work as well (including subscription etc.) Use the very latest - 6.2.1. It seems quite good. Earlier versions (including 6.2.0) had problems. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
TWV wrote: By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware I had to revert back to 5.5, because 6.0 kept "garbling" my LCD screen (the screen would become unreadable). You might want to wait for 6.0.1 :-) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
I'm sorry to hear that, but I didn't experience such a problem, 6.0 seems to work quite well on my phone. Do you have a suggestion for my question? Or alternative: Is it possible to send a custom SIP NOTIFY message (with XML body) to an asterisk sip client? - Frederic -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Dr. Michael J. Chudobiak Verzonden: maandag 17 april 2006 18:45 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser TWV wrote: > By now, every Snom fan should have installed the 6.0 (beta) firmware :-) > See http://www.snom.com/wiki/index.php/Beta_Firmware I had to revert back to 5.5, because 6.0 kept "garbling" my LCD screen (the screen would become unreadable). You might want to wait for 6.0.1 :-) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
