[asterisk-users] Local Channel Resource Limit
Hello, How can I set a hard limit to the number of Local channels asterisk can spawn? -- Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP Questions
Greetings Ron- Just wanted to give you a heads up about an alternative SCCP channel driver available for Asterisk. Please see here: http://freecode.com/projects/chan-sccp-b I have no experience with it (nor SCCP in general) but just wanted to give you an option in the event the included SCCP driver does not give you satisfactory results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local Channel Resource Limit
On 06/14/2012 04:20 AM, [Digital^Dude] ® wrote: How can I set a hard limit to the number of Local channels asterisk can spawn? chan_local does not have a mechanism to do this. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom, Dial Specific Number on Handset Pickup
http://lists.digium.com/pipermail/asterisk-users/2012-February/270427.html That worked for me with the polycom 3.x firmware; I haven't tried it with 4.0 firmware yet. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: Wednesday, June 13, 2012 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion (asterisk-users@lists.digium.com) Subject: [asterisk-users] Polycom, Dial Specific Number on Handset Pickup Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with ss7 voice broadcast
Hello, Asterisk under 90% load of SS7 calls can only withstand the voice broadcasting for 30 minutes. After around 30 minutes, it stops receiving any call hits via AMI. No errors are reported. Giving it a minute's rest makes it work for another 30 minutes. Can anyone hint to what may be causing this? -- Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dualstack
I have an Asterisk (v10.2.0) running and bound to address ::. I think this way he listens and answers to requests send to the IPv4 and IPv6 address (haven't check that with IPv6 yet). What I want to achieve is, that he handles signaling via IPv4, but RTP via IPv6. In my setup, I have a user agent (Dualstack) that generates an INVITE and sends it out via IPv6. In the SDP part the user agent expects the RTP traffic on its IPv6 address as well. Between the user agent and the Asterisk, I have a proxy that handles the signaling part, and translates from IPv6 to IPv4 and vice versa. The Asterisk accepts that request (IPv4) and does everything well, except that in his SDP offer, he inserts his IPv4 address (I think that's because he received the request via IPv4). The result of this is: The user agent sends RTP traffic via IPv4 to the Asterisk. The Asterisk sends RTP traffic via IPv6 to the user agent. Signaling: UA (IPv6) Proxy (IPv4) Asterisk RTP: UA --- (IPv4) Asterisk UA - (IPv6) -- Asterisk Does anybody know how I can achieve that Asterisk does input his IPv6 address in the SDP offer and uses that for incoming RTP, if he sees, that the user agent also uses an IPv6 address in his SDP offer? Maybe there is an easy way and I've just overseen a configuration option. Or do I have to patch the sources? Thanks in advance! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dualstack
- Original Message - From: Pawel Kuzak pawel.ku...@1und1.de To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 14, 2012 10:22:21 AM Subject: [asterisk-users] Dualstack I have an Asterisk (v10.2.0) running and bound to address ::. I think this way he listens and answers to requests send to the IPv4 and IPv6 address (haven't check that with IPv6 yet). What I want to achieve is, that he handles signaling via IPv4, but RTP via IPv6. In my setup, I have a user agent (Dualstack) that generates an INVITE and sends it out via IPv6. In the SDP part the user agent expects the RTP traffic on its IPv6 address as well. Between the user agent and the Asterisk, I have a proxy that handles the signaling part, and translates from IPv6 to IPv4 and vice versa. The Asterisk accepts that request (IPv4) and does everything well, except that in his SDP offer, he inserts his IPv4 address (I think that's because he received the request via IPv4). The result of this is: The user agent sends RTP traffic via IPv4 to the Asterisk. The Asterisk sends RTP traffic via IPv6 to the user agent. Signaling: UA (IPv6) Proxy (IPv4) Asterisk RTP: UA --- (IPv4) Asterisk UA - (IPv6) -- Asterisk Does anybody know how I can achieve that Asterisk does input his IPv6 address in the SDP offer and uses that for incoming RTP, if he sees, that the user agent also uses an IPv6 address in his SDP offer? Maybe there is an easy way and I've just overseen a configuration option. Or do I have to patch the sources? Thanks in advance! Assuming that the same address can be sent to all user agents that communicate with that Asterisk instance, the media_address setting could potentially be used to specify an IPv6 address to send media to, while keeping the signalling on IPv4. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP Questions
Hello, Thanks you for the replies ill take a look at the driver you sent over. Im going to run some test and see what happens, hopefully the driver in 1.8 is soild and nothing needs to be messed with, but we will see :) On Thu, Jun 14, 2012 at 5:06 AM, Tim Nelson tnel...@rockbochs.com wrote: Greetings Ron- Just wanted to give you a heads up about an alternative SCCP channel driver available for Asterisk. Please see here: http://freecode.com/projects/chan-sccp-b I have no experience with it (nor SCCP in general) but just wanted to give you an option in the event the included SCCP driver does not give you satisfactory results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differences between PBX and SBC
That's my question...the sbc provides security over trunking, right? The same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of add-value to an Asterisk deployment? A PBX provides functionality to users. An SBC *can* secure a PBX against the outside world, but that is configuration dependent. The more powerful the SBC, the more configuration it requires to make things work, let alone secure whatever it is supposed to protect. An SBC is in essence a B2BUA, looking quote a lot like a really simple pass through Asterisk configuration. -- Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SCCP Questions
Is the chan-sccp-b project the same one that got put in SVN of 1.8 branch? I have not been able to find anything definitive that says so, I really need 1.8 branch so trying to see which is the best way to go. Thanks On Thu, Jun 14, 2012 at 9:34 AM, Ron McCarthy ronmc...@gmail.com wrote: Hello, Thanks you for the replies ill take a look at the driver you sent over. Im going to run some test and see what happens, hopefully the driver in 1.8 is soild and nothing needs to be messed with, but we will see :) On Thu, Jun 14, 2012 at 5:06 AM, Tim Nelson tnel...@rockbochs.com wrote: Greetings Ron- Just wanted to give you a heads up about an alternative SCCP channel driver available for Asterisk. Please see here: http://freecode.com/projects/chan-sccp-b I have no experience with it (nor SCCP in general) but just wanted to give you an option in the event the included SCCP driver does not give you satisfactory results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.5.1 Now Available (Security Release)
The Asterisk Development Team has announced a security release for Asterisk 10. This security release is released as version 10.5.1. The release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 10.5.1 resolves the following issue: * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) Channel driver. When an SCCP client sends an Off Hook message, followed by a Key Pad Button Message, a structure that was previously set to NULL is dereferenced. This allows remote authenticated connections the ability to cause a crash in the server, denying services to legitimate users. This issue and its resolution is described in the security advisory. For more information about the details of this vulnerability, please read security advisory AST-2012-009, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.1 The security advisory is available at: * http://downloads.asterisk.org/pub/security/AST-2012-009.pdf Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2012-009: Skinny Channel Driver Remote Crash Vulnerability
Asterisk Project Security Advisory - AST-2012-009 Product Asterisk Summary Skinny Channel Driver Remote Crash Vulnerability Nature of Advisory Denial of Service Susceptibility Remote authenticated sessions SeverityMinor Exploits Known No Reported On May 30, 2012 Reported By Christoph Hebeisen, TELUS Security Labs Posted OnJune 14, 2012 Last Updated On June 14, 2012 Advisory ContactMatt Jordan mjordan AT digium DOT com CVE NameCVE-2012-3553 Description AST-2012-008 previously dealt with a denial of service attack exploitable in the Skinny channel driver that occurred when certain messages are sent after a previously registered station sends an Off Hook message. Unresolved in that patch is an issue in the Asterisk 10 releases, wherein, if a Station Key Pad Button Message is processed after an Off Hook message, the channel driver will inappropriately dereference a Null pointer. Similar to AST-2012-008, a remote attacker with a valid SCCP ID can can use this vulnerability by closing a connection to the Asterisk server when a station is in the Off Hook call state and crash the server. Resolution The presence of a device for a line is now checked in the appropriate channel callbacks, preventing the crash. Affected Versions Product Release Series Asterisk Open Source 10.x All Versions Corrected In Product Release Asterisk Open Source10.5.1 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2012-009-10.diff v10 Links https://issues.asterisk.org/jira/browse/ASTERISK-19905 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2012-009.pdf and http://downloads.digium.com/pub/security/AST-2012-009.html Revision History Date Editor Revisions Made 06/14/2012 Matt Jordan Initial Release Asterisk Project Security Advisory - AST-2012-009 Copyright (c) 2012 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium IP Phones - Teleworker Capability?
We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good teleworker functionality? The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can log in to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Thanks for all comments! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On 06/14/2012 04:57 PM, asterisk users wrote: We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good teleworker functionality? Yes, I believe they do :-) The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. Digium phones can do something similar, and in an upcoming firmware release, there will even be features available to make this happen on a fairly automatic basis. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. Digium phones speak SIP and RTP to the server, just like pretty much any other SIP phone. They employ many modern NAT traversal techniques and should work in most network situations. They don't currently provide encryption for signaling and media, though. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. I don't understand this; SIP phones don't require this at all. The phone is an intelligent device on its own. If there is no network connectivity to the server, then calls cannot be placed or received, but once connectivity is restored, operation would be back to normal. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can log in to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). Yes, this is supported. These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Most of it, I think. Give them a try! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/14/2012 04:57 PM, asterisk users wrote: We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good teleworker functionality? Yes, I believe they do :-) The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. Digium phones can do something similar, and in an upcoming firmware release, there will even be features available to make this happen on a fairly automatic basis. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. Digium phones speak SIP and RTP to the server, just like pretty much any other SIP phone. They employ many modern NAT traversal techniques and should work in most network situations. They don't currently provide encryption for signaling and media, though. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. I don't understand this; SIP phones don't require this at all. The phone is an intelligent device on its own. If there is no network connectivity to the server, then calls cannot be placed or received, but once connectivity is restored, operation would be back to normal. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can log in to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). Yes, this is supported. These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Most of it, I think. Give them a try! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- This is pretty good news, overall. To comment on Kevin's points: - The end-to-end encryption is important to us, because client-ID-sensitive information is part of our environment. Something like built-in OpenVPN would work for us, if that were an option. - Being fault-tolerant (of less than perfect DSL and rural-wireless connections - if the boss is at his cabin, for instance) and being very user-friendly about it is really important to end users. Minet has a heart-beat mechanism so that if the connection goes down between the phone and the switch, the display shows it. Of course, calls get diverted to voicemail during that period. If something is not working in the network, the user is informed about it, and when it is fixed, everything continues, including button DSS status updates, voicemail WMI, etc. On typical SIP phones, everything looks normal until you go to use it, then there is no dialtone, or you just get dead-air on the handset). Our users are pretty demanding, and want a utility-grade solution that will always work - for them. - Most of it, I think. Give them a try! Is there a detailed application note in the Digium wiki (or anywhere else for that matter) about these implementing features under Asterisk/Switchvox? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On Thu, 2012-06-14 at 16:23 -0600, asterisk users wrote: This is pretty good news, overall. To comment on Kevin's points: - The end-to-end encryption is important to us, because client-ID-sensitive information is part of our environment. Something like built-in OpenVPN would work for us, if that were an option. Yealink and I think Aastra phones have OpenVPN built in. We use Yealink with layer 2 tunnels such that the phones have the same configuration, network wise, wherever they happen to be plugged in. No NAT issues ever. - Being fault-tolerant (of less than perfect DSL and rural-wireless connections - if the boss is at his cabin, for instance) and being very user-friendly about it is really important to end users. Minet has a heart-beat mechanism so that if the connection goes down between the phone and the switch, the display shows it. Of course, calls get diverted to voicemail during that period. Pretty much all SIP phones work that way. If something is not working in the network, the user is informed about it, and when it is fixed, everything continues, including button DSS status updates, voicemail WMI, etc. Again all phones work that way. On typical SIP phones, everything looks normal until you go to use it, then there is no dialtone, or you just get dead-air on the handset). Which SIP phone have you been using? The ones we are familiar with - Polycom, Linksys, Yealink, Snom, Aastra, Grandstream - all show you when the network link is down, and all services return as soon as it comes back up. Even Linksys ATAs at least show you an LED of when the device is registered, though you will just get dead air if you pick up the handset. Our users are pretty demanding, and want a utility-grade solution that will always work - for them. - Most of it, I think. Give them a try! Is there a detailed application note in the Digium wiki (or anywhere else for that matter) about these implementing features under Asterisk/Switchvox? You could probably find 50 people to help you set such a system up on this list (or more appropriately on -biz). Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom, Dial Specific Number on Handset Pickup
On Thu, 14 Jun 2012 00:46:25 + Klaverstyn, David C david.klavers...@intergraph.com wrote: I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. call.autoOffHook call.autoOffHook.1.contact= call.autoOffHook.1.enabled=1 call.autoOffHook.1.protocol= /call.autoOffHook -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk support AMR and AMR-WB
Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which support AMR, AMR-WB recommended are welcome. Best regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
