Re: [asterisk-users] Snom 360 lights not working on subscription / fail to extend xx to xx error
Dear guys, many people have been using Snom with Subscription/notify lights I tried almost every tip in the google. But there's one thing related to the snom phones and asterisk I didn't see in any forum The Asterisk console show very often a message like: fail to extend from xx to xxx This message appears ver often and when snom phones do reboot or subscribe or while it receives notify messages. Any idea? BTW. in Snom's sip trace content length is 0 in every NOTIFY message received by phone and there's no XML thanks and regards, Carlos On Oct 23, 2007, at 8:55 PM, Craig Guy wrote: The Linksys SPA962 with SPA932 sidecar support both speed dial and BLF. IMHO very good for the money and very easy to provision once you get a hold of the proper provisioning guide. These things are designed for mass deployment and remote provisioning. As other people have noted, you need to provision via http rather than tftp for best effect. I also have two provisioning files, a shared settings file with the bulk of the config and then a per handset file based on the mac address containing the account and any special customisations. The only bad bit is that a resync usually causes a reboot of the handset which interrupts the connection of anything attached to the PC port of the phone. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Tuesday, 23 October 2007 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription Hey Mike, We started deploying exclusively Polycom and Linksys. The Polycom's support presence, they call it 'Buddy List'. I am not sure about the Linksys phones, I don't think they do although I did see support for SLA (Shared Line Appearance). Omar On 10/23/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options
Re: [asterisk-users] Snom 360 lights not working on subscription
In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. What we found is that even if you get the lights working, they go off after a few days. The BLF lights on the Snom 360s work for me (Asterisk 1.4, Snom 6.5.12 firmware), but I reboot them nightly. I have noticed that the Snom BLFs can stop working if the network is busy for a long period of time (i.e., longer than the re-registration period), like during system-wide backups and yum-upgrades. To avoid this problem, I have a cron job reboot the Snoms nightly after scheduled backups/upgrades. I'm not sure if this is a network congestion issue or a server CPU overload issue, or something else. Anyway, this arrangement does seem to be pretty reliable. To reboot a Snom: http://www.voip-info.org/wiki/view/Asterisk+phone+snom#RebootingaSNOM360320. Hope this helps. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
The Linksys SPA962 with SPA932 sidecar support both speed dial and BLF. IMHO very good for the money and very easy to provision once you get a hold of the proper provisioning guide. These things are designed for mass deployment and remote provisioning. As other people have noted, you need to provision via http rather than tftp for best effect. I also have two provisioning files, a shared settings file with the bulk of the config and then a per handset file based on the mac address containing the account and any special customisations. The only bad bit is that a resync usually causes a reboot of the handset which interrupts the connection of anything attached to the PC port of the phone. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Tuesday, 23 October 2007 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription Hey Mike, We started deploying exclusively Polycom and Linksys. The Polycom's support presence, they call it 'Buddy List'. I am not sure about the Linksys phones, I don't think they do although I did see support for SLA (Shared Line Appearance). Omar On 10/23/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 360 lights not working on subscription
Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
What we found is that even if you get the lights working, they go off after a few days. Paul Hales AsteriskIT On Mon, 2007-10-22 at 09:49 -0300, Carlos Maimone wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
Hey Mike, We started deploying exclusively Polycom and Linksys. The Polycom's support presence, they call it 'Buddy List'. I am not sure about the Linksys phones, I don't think they do although I did see support for SLA (Shared Line Appearance). Omar On 10/23/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM 360 Rejecting Calls
Does someone know coincidentally the cause for the error message specified in the Subject? The following scenario: Snom 360 behind one rout (wiederrum on a DSL line with static IP address hangs). The Snom has a private IP, routs accomplishes NAT. STUN and ICE are activated, as SIP haven 5060/udp are firmly used. Detailed packages passed on on haven 5060/udp of rout to the Snom. The telephone 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? ___ --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
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 asterisk-users@lists.digium.com 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
[asterisk-users] SNOM 360
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
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
[asterisk-users] Snom 360 Caller ID in missed / recieved calls
Hi List, We have noticed on our Snom 360s that under missed/recieved calls the number is cut off, so you cannot see the entire phone number. Does anyone have a work around or is this a bug Snom is working on? Cheers! ___ --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
[asterisk-users] SNOM 360
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
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: http://somesite/ringer.wav) 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=getsearch=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 ___ --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
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($[ ${CALLERID(number)} =
2101 ]?skipself)
exten = PAGE2101,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2101,n,Dial(SIP/2101,5)
exten = PAGE2101,n(skipself),Noop(Not paging
originator)
exten = PAGE2102,1,GotoIf($[ ${CALLERID(number)} =
2102 ]?skipself)
exten = PAGE2102,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2102,n,Dial(SIP/2102,5)
exten = PAGE2102,n(skipself),Noop(Not paging
originator)
exten = PAGE2105,1,GotoIf($[ ${CALLERID(number)} =
2105 ]?skipself)
exten = PAGE2105,n,Set(__SIPADDHEADER=Call-Info:
\;answer-after=0)
exten = PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
exten =
PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2105,n,Dial(SIP/2105,5)
exten = PAGE2105,n(skipself),Noop(Not paging
originator)
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]
aging)
exten =
99,1,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/PAGE
[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${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/${EXTEN:2})
__
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[asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${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/${EXTEN:2})
__
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RE: [asterisk-users] snom 360 auto answer
This is my code (that I copied form somewhere) for paging a group of
phones. By dialling 99 it will page phones 2101, 2102 and 2105.
Just include the context ext-paging in your dial plan and modify the
extension numbers and all should be good.
This works on Linksys Phones but should also work on Snoms.
I hope this helps you.
[ext-paging]
exten = PAGE2101,1,GotoIf($[ ${CALLERID(number)} = 2101 ]?skipself)
exten = PAGE2101,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten = PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2101,n,Dial(SIP/2101,5)
exten = PAGE2101,n(skipself),Noop(Not paging originator)
exten = PAGE2102,1,GotoIf($[ ${CALLERID(number)} = 2102 ]?skipself)
exten = PAGE2102,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten = PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2102,n,Dial(SIP/2102,5)
exten = PAGE2102,n(skipself),Noop(Not paging originator)
exten = PAGE2105,1,GotoIf($[ ${CALLERID(number)} = 2105 ]?skipself)
exten = PAGE2105,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten = PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE2105,n,Dial(SIP/2105,5)
exten = PAGE2105,n(skipself),Noop(Not paging originator)
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]
aging)
exten =
99,1,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]LOCAL/PAGE
[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer
Hi,
I'm testing paging using snom 360.
Can someone correct my dialplan?
Regards,
Jason.
==
;exten = _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten = _99,n,SIPAddHeader(Call-Info:
sip:192.168.1.113\;answer-after=0)
;exten = _99,n,Dial(SIP/${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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[asterisk-users] Snom 360 / firmware 6.5.1 / registration problems with Asterisk
Hi, we just upgraded from 1.2.10 to 1.2.13 and now encounter strange problems with our snom phones (FW 6.2.3 to 6.5.1). Upon phone boot everything works fine. Phone registers and asterisk is happy. Soon afterwards the registration is lost however. Sometimes after a few minutes the phone reregisters, sometimes not. This only seems to happen on the first configured line. Switching back to 1.2.10 solved the problem. What changed between those to versions? Maybe a new setting on the snoms we have to take care of? Funny thing: I set defaultexpiry=60 and told the phone to use 1min as well. After the phone registered I watched the expiry counter with sip show peer. It counted backwards from 60 to about 40. Then it jumped to 70, counted to 0 and the phone was gone. This is somewhat reproducable. And it simply does not look right... Any ideas? Kind 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
[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
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 asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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: how to make record button working ?
Jay R. Ashworth wrote: So, you're suggesting that the FXO channel driver generates outbound DTMF under the command of (eventually) the phone set? That would be nice. Yes, that _would_ be nice. Are you suggesting that that's not what's happening? I'm not sure I gather your meaning, or I could be incorrectly discerning sarcasm. I tried to disclaim my ignorance _and_ answer Remco's concern regarding DTMF reaching remote IVRs. When I press the monitor sequence on my phone, the remote party doesn't hear anything, not even a crackle. Same with the blind transfer sequence. Moj ___ --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: how to make record button working ?
On Mon, Oct 09, 2006 at 08:50:51AM -0800, Mojo with Horan Company, LLC wrote: Jay R. Ashworth wrote: So, you're suggesting that the FXO channel driver generates outbound DTMF under the command of (eventually) the phone set? That would be nice. Yes, that _would_ be nice. Are you suggesting that that's not what's happening? I'm not sure I gather your meaning, or I could be incorrectly discerning sarcasm. I tried to disclaim my ignorance _and_ answer Remco's concern regarding DTMF reaching remote IVRs. You were incorrectly discerning sarcasm. Sorry. I was hoping it worked as described. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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: how to make record button working ?
On Fri, Oct 06, 2006 at 08:35:26AM -0800, Mojo with Horan Company, LLC wrote: I'm pretty sure that when you AREN'T sending the DTMF inband, asterisk detects it, and if the keys pressed don't lead to any recording/transfer features, then it re-creates DTMF on the bridged channel. I mean to say, my called party can't hear me start recording or transfer them, but I don't have any trouble with outside IVRs. So, you're suggesting that the FXO channel driver generates outbound DTMF under the command of (eventually) the phone set? That would be nice. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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: how to make record button working ?
I cannot find this option in the snom firmware, the only thing I found is DTMF via SIP INFO: This sounds nice but I guess it will break stuff if you need DTMF tones to get through the menu of a remote PBX. Ideally * would need to interpret the SIP INFO message from the Snom as start recording. I looked at the patch someone mentioned earlier but to me this looks like re-inventing the wheel by starting the whole recording stuff all over again. All this is not necessary, * should simply treat the SIP INFO message the same as DTMF dialling *1 On Thu, 5 Oct 2006, Mojo with Horan Company, LLC wrote: We use SIP Polycom 501s, and their dtmfmode=rfc2833. The remote party can NOT hear the tones when you start recording. I suspect that if dtmfmode=inband, they WOULD be able to. Could be wrong here, that's just my current rudimentary understanding of the situation :) Moj Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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 !DSPAM:500,452494b8123922068143078! ___ --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: how to make record button working ?
Another way would be to set the dtmf option to speed dial and then add a speed dial number 1: *1 ___ --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: how to make record button working ?
Remco Barendse wrote: I cannot find this option in the snom firmware, the only thing I found is DTMF via SIP INFO: This sounds nice but I guess it will break stuff if you need DTMF tones to get through the menu of a remote PBX. I'm pretty sure that when you AREN'T sending the DTMF inband, asterisk detects it, and if the keys pressed don't lead to any recording/transfer features, then it re-creates DTMF on the bridged channel. I mean to say, my called party can't hear me start recording or transfer them, but I don't have any trouble with outside IVRs. Ideally * would need to interpret the SIP INFO message from the Snom as start recording. I looked at the patch someone mentioned earlier but to me this looks like re-inventing the wheel by starting the whole recording stuff all over again. All this is not necessary, * should simply treat the SIP INFO message the same as DTMF dialling *1 On Thu, 5 Oct 2006, Mojo with Horan Company, LLC wrote: We use SIP Polycom 501s, and their dtmfmode=rfc2833. The remote party can NOT hear the tones when you start recording. I suspect that if dtmfmode=inband, they WOULD be able to. Could be wrong here, that's just my current rudimentary understanding of the situation :) Moj Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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 !DSPAM:500,45261e8d254852002735277! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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: how to make record button working ?
On Thu, 5 Oct 2006, Joel Hill wrote: No worries. Good question, I wasn't sure so I just tested it and it seems that the answer is yes it does send the tones to the other side. Can I ask why this would matter, I think there could be legal implications of recording a call and not notifying the other party. That's why you always get the message This call may be monitored for training and coaching purposes. Etc.. AFAIK in The Netherlands there is no law to obligatory tell the other party that the conversation is / will be recorded, banks do it as a standard procedure for example when you are placing forex orders. I think even insurance companies do the same when you call in to report a claim/damage. With audible DTMF tones this function is basically unusable for our purposes. With the recording function already being implemented in * I guess it would be trivial to get it working with the SIP info message as well? (Or maybe it is intentional behaviour that it will only work out of the box with audible DTMF tones) Cheers! Remco ___ --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: how to make record button working ?
That varies from location to location, really. In Georgia, for instance, only ONE party need know the recording is taking place (calling or receiving) without a warrant. In some countries, neither party need know, etc, etc. N. On Thu, 05 Oct 2006 15:25:28 +1000, Joel Hill wrote No worries. Good question, I wasn't sure so I just tested it and it seems that the answer is yes it does send the tones to the other side. Can I ask why this would matter, I think there could be legal implications of recording a call and not notifying the other party. That's why you always get the message This call may be monitored for training and coaching purposes. Etc.. Cheers, Joel. Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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 ___ --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: how to make record button working ?
There was a patch to get this working, looks like it has been abandoned, though. Should give you a starting point to get it working, or perhaps a bounty would get someone interested in getting it usable and committed. http://bugs.digium.com/view.php?id=4845 On 10/4/06, Joel Hill [EMAIL PROTECTED] wrote: Hi Noro,Depending on what firmware you have this is the way to go.Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asteriskrecording function.Hope this helpsCheers,JoelAsterisk ITwww.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output,the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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: how to make record button working ?
Hi Joel, thanks for the answer :-). Yes this is one (the easiest) way how it can be done (on phone side), but I am still looking for asterisk side solution ... i.e. it should understand info message sent by phone and do some prescribed action. Haven't u any clue ? noro 2006/10/5, Joel Hill [EMAIL PROTECTED]: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ ___ --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: how to make record button working ?
On Thu, Oct 05, 2006 at 07:17:47AM -0400, sip wrote: That varies from location to location, really. In Georgia, for instance, only ONE party need know the recording is taking place (calling or receiving) without a warrant. In some countries, neither party need know, etc, etc. This page: http://www.pimall.com/nais/n.recordlaw.html purports to list the states that require all party consent. It is from a private investigation site, and was the number one google hit, so it may be reliable. This is not legal advice; IANAL. If my advice breaks something, you get to keep both pieces, unless you paid me for it. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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: how to make record button working ?
Hi Joe, this is the link I was looking for - I ggled a lot, but didn't find it. Thanke you ! noro 2006/10/5, Joe Pukepail [EMAIL PROTECTED]: There was a patch to get this working, looks like it has been abandoned, though. Should give you a starting point to get it working, or perhaps a bounty would get someone interested in getting it usable and committed. http://bugs.digium.com/view.php?id=4845 On 10/4/06, Joel Hill [EMAIL PROTECTED] wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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: how to make record button working ?
We use SIP Polycom 501s, and their dtmfmode=rfc2833. The remote party can NOT hear the tones when you start recording. I suspect that if dtmfmode=inband, they WOULD be able to. Could be wrong here, that's just my current rudimentary understanding of the situation :) Moj Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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 !DSPAM:500,452494b8123922068143078! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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
[asterisk-users] snom 360: how to make record button working ?
Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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: how to make record button working ?
Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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
[asterisk-users] snom 360 - how to make record button working ?
Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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: how to make record button working ?
Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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: how to make record button working ?
No worries. Good question, I wasn't sure so I just tested it and it seems that the answer is yes it does send the tones to the other side. Can I ask why this would matter, I think there could be legal implications of recording a call and not notifying the other party. That's why you always get the message This call may be monitored for training and coaching purposes. Etc.. Cheers, Joel. Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --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 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: sip:[EMAIL PROTECTED];user=phone 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
[asterisk-users] Snom 360 Function Keys
I have a Snom 360 phone and I'm configuring it for use with Asterisk 1.2.9 and Freepbx 2.1.1 On my PBX there are: 1) Some SIP phones 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? Configure sip phones is very simple (just put in function keys panel the SIP URI of every phone) but I have same problems with analogue phones! Someone have the same problem? ___ --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 asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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 quickdirty 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
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
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 asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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 asterisk-users@lists.digium.com 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
[asterisk-users] SNOM 360
Hi List,Does anyone know how to set up QoS on the SNOM 360 ? 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
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
[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
[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.) Kind regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --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 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 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 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
[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
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
[Asterisk-Users] Snom 360 doesn't register after reboot
Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click Re-register in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani ___ --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 doesn't register after reboot
Mimmus wrote: Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click Re-register in the web interface. I think that was fixed in 6.2.1. See http://www.snom.com/wiki/index.php/Beta_Firmware and http://www.voip-info.org/wiki/view/snom+360 - 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
Hello, why don't you send a menu list back and the user has to select a item and he browse to a certain url? If not then we have to implement that. Is that so important for you? No problem to make a firmware version, let discuss why do you need this... Suggestion: SnomIPPhoneRedirect urlhttp://example.org/url /SnomIPPhoneRedirect best regards, Hirosh Dabui TWV wrote: Dear Hirosh, I already knew about that :-), and tried it with success. However, that was not my question! I asked how you can make the phone browse to a certain http:// URL, initiated from the server side. So essentially remote control the phone to open an XML file from some server in its microbrowser! Is there a specific SIP NOTIFY for that too? In the example that you link too, the XML is provided as body of the notify message itself. I only want to send an existing URL to the phone. I hope you have thought about this functionality, it would be very useful! Thank you very much, Frederic -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Hirosh Dabui Verzonden: dinsdag 18 april 2006 17:02 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser Hello, snom 360s can handle xml messages via SIP-Notify. Descriptions how to implement this on: http://snom.com/minibrowser/doc/xmlapplsnom360.pdf http://snom.com/minibrowser/notify.txt Common infos you can find out on: http://snom.com/wiki/index.php/Xmlobjects Hope this will help... cheers, Hirosh 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 The XML minibrowser is very cool and opens a lot of possibilities! One of my ideas is rich messaging, so you can send fully formatted messages to a Snom 360 user! But... how can you make the phone navigate to a certain URL? (Initiated from the Asterisk side of course!) Is there some sort of SIP message or Asterisk Application / Command that can be used to make the phone browse to an xml URL? If not, this is a call to the nice people of Snom or the Asterisk community to add this functionality, it will be much needed! Thanks, Frederic ___ --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 Hirosh Dabui PGP Key-ID: 0x30A34758 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 Firmware 6.0 - Microbrowser
Hello, snom 360s can handle xml messages via SIP-Notify. Descriptions how to implement this on: http://snom.com/minibrowser/doc/xmlapplsnom360.pdf http://snom.com/minibrowser/notify.txt Common infos you can find out on: http://snom.com/wiki/index.php/Xmlobjects Hope this will help... cheers, Hirosh 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 The XML minibrowser is very cool and opens a lot of possibilities! One of my ideas is rich messaging, so you can send fully formatted messages to a Snom 360 user! But... how can you make the phone navigate to a certain URL? (Initiated from the Asterisk side of course!) Is there some sort of SIP message or Asterisk Application / Command that can be used to make the phone browse to an xml URL? If not, this is a call to the nice people of Snom or the Asterisk community to add this functionality, it will be much needed! Thanks, Frederic ___ --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 Hirosh Dabui PGP Key-ID: 0x30A34758 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 Firmware 6.0 - Microbrowser
Dear Hirosh, I already knew about that :-), and tried it with success. However, that was not my question! I asked how you can make the phone browse to a certain http:// URL, initiated from the server side. So essentially remote control the phone to open an XML file from some server in its microbrowser! Is there a specific SIP NOTIFY for that too? In the example that you link too, the XML is provided as body of the notify message itself. I only want to send an existing URL to the phone. I hope you have thought about this functionality, it would be very useful! Thank you very much, Frederic -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Hirosh Dabui Verzonden: dinsdag 18 april 2006 17:02 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser Hello, snom 360s can handle xml messages via SIP-Notify. Descriptions how to implement this on: http://snom.com/minibrowser/doc/xmlapplsnom360.pdf http://snom.com/minibrowser/notify.txt Common infos you can find out on: http://snom.com/wiki/index.php/Xmlobjects Hope this will help... cheers, Hirosh 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 The XML minibrowser is very cool and opens a lot of possibilities! One of my ideas is rich messaging, so you can send fully formatted messages to a Snom 360 user! But... how can you make the phone navigate to a certain URL? (Initiated from the Asterisk side of course!) Is there some sort of SIP message or Asterisk Application / Command that can be used to make the phone browse to an xml URL? If not, this is a call to the nice people of Snom or the Asterisk community to add this functionality, it will be much needed! Thanks, Frederic ___ --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 Hirosh Dabui PGP Key-ID: 0x30A34758 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 ___ --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
[Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware The XML minibrowser is very cool and opens a lot of possibilities! One of my ideas is rich messaging, so you can send fully formatted messages to a Snom 360 user! But... how can you make the phone navigate to a certain URL? (Initiated from the Asterisk side of course!) Is there some sort of SIP message or Asterisk Application / Command that can be used to make the phone browse to an xml URL? If not, this is a call to the nice people of Snom or the Asterisk community to add this functionality, it will be much needed! Thanks, Frederic ___ --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
Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
In Cisco land you would send a command to the phone via a long URL so the idea was to send a HTTP/POST to ipaddress/ciscoservices/directory/browser/index.html?url=http://somedomain/services/app.php this is not exact as it changes often. I am reading on the snoms I am sure their system is much better than Cisco's in the openess department :) On 4/17/06, TWV [EMAIL PROTECTED] wrote: 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 -- --- 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! 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 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
[Asterisk-Users] Snom 360 - Multiple Server BLF Indications
Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart begin:vcard fn:Stuart Elvish n:Elvish;Stuart org:Dallas Delta Corporation Pty Ltd;Voice Networking Directorate adr:;;102 Albert Street;East Brunswick;VIC;3057;Australia email;internet:[EMAIL PROTECTED] title:Voice Networking Engineer tel;work:03 9387 7445 tel;fax:03 9387 3128 tel;cell:0408 873 601 x-mozilla-html:TRUE url:http://www.dallasdelta.net version:2.1 end:vcard ___ --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 - Multiple Server BLF Indications
I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart ___ --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 - Multiple Server BLF Indications
I installed 2 Snom360's a few months ago, and 'at the time' only 1 expansion module could be added. (also the fact that the modules draw so much current that it got the POE switch upset!) Have you tested a snom360? I should have one in the lab soon enough. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 11:41 AM Subject: Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications There is an add on module for this phone and according to a source that distributes them here, the modules can be daisy chained until you reach the required number of extensions. I didn't think you could, but that is the information that we have at hand... [EMAIL PROTECTED] wrote: I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart -- -- ___ --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 - Multiple Server BLF Indications
I had a look at the snom website - and the manual for the expansion module read that only one module can be attached 'currently'. So maybe this has changed. Any ideas? Personally, I like snom phones a lot. I used a snom 200 at my desk at a previous job for almost 2 years. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 11:41 AM Subject: Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications There is an add on module for this phone and according to a source that distributes them here, the modules can be daisy chained until you reach the required number of extensions. I didn't think you could, but that is the information that we have at hand... [EMAIL PROTECTED] wrote: I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart -- -- ___ --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 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
[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? 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
[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 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? 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
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:
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 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 problem - only one call works after reboot
After rebooting, I can make one outgoing call successfully. Subsequent calls don't work - the 360 just seems to do nothing after pressing the OK button (but I can cancel the call, the phone isn't frozen). The Asterisk console shows the first call going through, but nothing appears for the subsequent calls, so they aren't even getting to Asterisk. Define an outbound proxy for your line. Dan, Thanks, but that wasn't the problem. I had to set RTP Encryption on the snom 360 to off. By default it is on. I have no idea why it causes a problem, but that is the solution! - 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
