Re: [asterisk-users] Can Asterisk handle in any way an SDP with m=application webrtc-datachannel ?
On Mon, Nov 21, 2016, at 03:06 PM, Alex Villacís Lasso wrote: > Is Asterisk capable of handling such a SDP, so that two SIP endpoints > registered through Asterisk can begin exchanging data? From what I > understand in the code, Asterisk will reject such a call. However, I want > to exhaust what Asterisk can do before > resorting to setting up a SIP proxy between the endpoint and Asterisk. I > remember reading a report about a webrtc success story where the > webphones were also exchanging data using datachannels. Asterisk does not support data channels and does not support exchanging SDP like this. A SIP proxy would be a better fit. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk help me with some requeriments, of my current project?
1 - My SIP server (Asterisk) will have some SIP clients registered in its SIP registrar. Let's say 6 SIP clients. In my project I have to implement a way of a SIP client making a call to a number and all others 5 SIP clients ring. That is, the others 5 SIP clients must receive the SIP INVITE. Can Asterisk help me with such functionality? The Dial() application lets you specify two or more destinations, separated by characters. When you execute an application call of this sort in your dialplan, Asterisk dials all of the destinations in parallel. If they're SIP clients, each will receive an INVITE at the same time. http://lists.digium.com/pipermail/asterisk-users/2005-April/094621.html 2 - When several SIP client ring, if one answer the call first, the others will have to stop ringing immediately. Can Asterisk help me with this requirement? If you use the dial in parallel technique I just described, when one client answers the call, Asterisk sends out a cancel invite to each of the other clients it had dialed. This *should* result in each of those other clients stopping their ring promptly... but that's up to the client. 3 - How to avoid one of the SIP clients receiving SIP INVITES? That is, one of the SIP clients is forbidden to receive calls. Is there a way to program it in Asterisk, maybe via dial plan? The question of which clients are called in response to a Dial() in your dial-plan, depends entirely on which clients are named in that Dial(). If you have five clients, and only include three of them in a particular Dial(), only those three will ring. If you have a client which is never named in a Dial() anywhere in your dialplan, Asterisk will never call it. It will be an outbound calls only client. 4- Let's suppose that I have a data base (let's say SQLite) in my SIP server (Asterisk) and I need implement a way of SIP Clients executing queries in such database. Could such queries be done/sent via SIP messages to Asterisk? Is there a way of accessing a database by meas of Asterisk, during a call, for example to collect information about others SIP Clients? Here I'm intending to create a software to be a kind of interface between Asterisk and the database, if necessary. In principle, a client could dial a URI which includes parameters for a SIP query. Asterisk's dialplan would recognize this URI (for example, it might start with *888* or some other such string), parse it, and feed the bits to an SQL query. With this approach (or any approach which accepts an SQL query or parameters from a client) you must be *EXTREMELY* careful to avoid SQL injection attacks. The story of little Bobby Tables is what I'm talking about here: https://xkcd.com/327/ 5 - If I need to send SIP messages all encrypted, using SSL or TLS , to the Asterisk, will this SIP server be able to interpret all messages correctly? Is there a way of let Asterisk talk with SIP clients in a secure way, using SSL, for example? Can Asterisk help me with this? https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial -- _ -- 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] can asterisk accept anonymous register ?
On Tue, Sep 14, 2010 at 5:59 AM, zhou tianjun zho...@gmail.com wrote: I want to know does the asterisk can realize that. Or I have to write module for that function ? No, you need to tell Asterisk what to do. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] Can asterisk work here
Yes, that should work fine, just remember you need a crossover cable to go from the a102 to the legacy system 2009/6/3 Jim Dickenson dicken...@cfmc.com I have a potential client that currently has a T1 circuit that feeds into an Adtran 750. Their phone sets are connected to the 24 ports on the 750. I was wondering if I could take an Asterisk system with a Sangoma A102de in it and plug the T1 into one port of the A102 and the 750 into the second port? Would I then have 24 voice channels that I could manage for the 24 phone sets? The only thing I know about the T1 is that it uses wink start signaling. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ ___ -- 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] Can asterisk work here
The only thing I know about the T1 is that it uses wink start signaling. Wink Start? That is an analog protocol used by DID or EM trunks. If that is what it is using, then the T1 must be a digitized set of DID analog trunks. A wink is a hook-switch-flash used to tell the originating side that it is ready to receive DTMF digits. Wilton ___ -- 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] Can asterisk work here
I think it is a DID trunk. I am having problems getting the clients telco to tell me much about the T1. For sure 24 analog channels in a single T1. Would I be able to use this type of T1 with a Sangoma A102de? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Wilton Helm wh...@compuserve.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jun 2009 11:09:31 -0600 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can asterisk work here The only thing I know about the T1 is that it uses wink start signaling. Wink Start? That is an analog protocol used by DID or EM trunks. If that is what it is using, then the T1 must be a digitized set of DID analog trunks. A wink is a hook-switch-flash used to tell the originating side that it is ready to receive DTMF digits. Wilton ___ -- 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] Can asterisk work here
On Wed, 3 Jun 2009, Jim Dickenson wrote: I think it is a DID trunk. I am having problems getting the clients telco to tell me much about the T1. For sure 24 analog channels in a single T1. Would I be able to use this type of T1 with a Sangoma A102de? -- Jim Dickenson mailto:dicken...@cfmc.com Yes, although I am having some amount of trouble with now TWO clients using this card with RBS T1's, which is what you probably have. Random unexplainable disconnects. Discussed earlier on this list I wasn't the first to have this problem, and others simply swapped to different hardware. My next step will be to try a Rhino T1 card, which is not only cheaper, but claims some hardware advantages over Sangoma/Digium. I would be very interested to hear if you have the same random hangup issue. Cheers, j CfMC http://www.cfmc.com/ From: Wilton Helm wh...@compuserve.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jun 2009 11:09:31 -0600 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can asterisk work here The only thing I know about the T1 is that it uses wink start signaling. Wink Start? That is an analog protocol used by DID or EM trunks. If that is what it is using, then the T1 must be a digitized set of DID analog trunks. A wink is a hook-switch-flash used to tell the originating side that it is ready to receive DTMF digits. Wilton ___ -- 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] Can asterisk work here
I finally got the provisioning for the T1. It is: T1 Service Type Robbed-Bit Signaling (RBS), four-wire Signal Protocol: EM Wink Line Coding: AMI Frame Mode: D4 Channels:24 That seems like something that the Sangoma card can support looking at web sites and such. I did see notes about the problem of rare mid conversation hang ups so I will watch for that. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Jeff LaCoursiere j...@jeff.net Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jun 2009 18:34:41 + (UTC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can asterisk work here On Wed, 3 Jun 2009, Jim Dickenson wrote: I think it is a DID trunk. I am having problems getting the clients telco to tell me much about the T1. For sure 24 analog channels in a single T1. Would I be able to use this type of T1 with a Sangoma A102de? -- Jim Dickenson mailto:dicken...@cfmc.com Yes, although I am having some amount of trouble with now TWO clients using this card with RBS T1's, which is what you probably have. Random unexplainable disconnects. Discussed earlier on this list I wasn't the first to have this problem, and others simply swapped to different hardware. My next step will be to try a Rhino T1 card, which is not only cheaper, but claims some hardware advantages over Sangoma/Digium. I would be very interested to hear if you have the same random hangup issue. Cheers, j CfMC http://www.cfmc.com/ From: Wilton Helm wh...@compuserve.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 3 Jun 2009 11:09:31 -0600 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can asterisk work here The only thing I know about the T1 is that it uses wink start signaling. Wink Start? That is an analog protocol used by DID or EM trunks. If that is what it is using, then the T1 must be a digitized set of DID analog trunks. A wink is a hook-switch-flash used to tell the originating side that it is ready to receive DTMF digits. Wilton ___ -- 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
Re: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call
Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes your life easy. -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased == Message: 16 Date: Fri, 10 Apr 2009 00:06:50 -0600 From: Shocky shoc...@users.sourceforge.net Subject: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server? To: asterisk-users@lists.digium.com Message-ID: 20090416.51201.shoc...@users.sourceforge.net Content-Type: text/plain; charset=us-ascii Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. Thanks, Shocky -- These are my opinions. Get your own. -- Message: 17 Date: Fri, 10 Apr 2009 10:07:38 +0300 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] MeetMe not working - was before To: asterisk-users@lists.digium.com Message-ID: 20090410070738.gs3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Apr 09, 2009 at 04:59:41PM -0400, John Rogers wrote: When I dial the extension of a meetme conference room, I get a message that states is not a valid conference. The meetme app was working before. I am getting this error on the CLI: app_meetme.c:800 build_conf: Unable to open pseudo device I have Asterisk 1.4.23.1 and zaptel-1.4.11 Elsewhere you mentioned you also have dahdi installed. What is the output of: ls /usr/include/dahdi I suspect Asterisk was built vs. dahdi whereas Zaptel was actually running. Actual tests: dahdi_test vs. zttest -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- Message: 18 Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST) From: Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net Subject: Re: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: pine.lnx.4.64.0904101032040.23...@unicorn.drogon.net Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 10 Apr 2009, Shocky wrote: Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. I've never looked at SCCP, but if it does work then you could use the console phone built into asterisk rather than IP plumb it into a soft-phone... So asterisk is essentially acting as an SCCP soft-phone itself. No GUI though
Re: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call
Sounds like the real question is: can Asterisk originate and receive SIP calls? The answer is yes. :-) -- Sent from mobile device On Apr 16, 2009, at 7:17 AM, Vidura Senadeera vidura...@gmail.com wrote: Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes your life easy. -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased == Message: 16 Date: Fri, 10 Apr 2009 00:06:50 -0600 From: Shocky shoc...@users.sourceforge.net Subject: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server? To: asterisk-users@lists.digium.com Message-ID: 20090416.51201.shoc...@users.sourceforge.net Content-Type: text/plain; charset=us-ascii Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. Thanks, Shocky -- These are my opinions. Get your own. -- Message: 17 Date: Fri, 10 Apr 2009 10:07:38 +0300 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] MeetMe not working - was before To: asterisk-users@lists.digium.com Message-ID: 20090410070738.gs3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Apr 09, 2009 at 04:59:41PM -0400, John Rogers wrote: When I dial the extension of a meetme conference room, I get a message that states is not a valid conference. The meetme app was working before. I am getting this error on the CLI: app_meetme.c:800 build_conf: Unable to open pseudo device I have Asterisk 1.4.23.1 and zaptel-1.4.11 Elsewhere you mentioned you also have dahdi installed. What is the output of: ls /usr/include/dahdi I suspect Asterisk was built vs. dahdi whereas Zaptel was actually running. Actual tests: dahdi_test vs. zttest -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- Message: 18 Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST) From: Gordon Henderson gordon+aster...@drogon.net Subject: Re: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: pine.lnx.4.64.0904101032040.23...@unicorn.drogon.net Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 10 Apr 2009, Shocky wrote: Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. I've never looked at SCCP, but if it does work then you could use the console phone built into asterisk rather than IP plumb
Re: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
On Fri, 10 Apr 2009, Shocky wrote: Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. I've never looked at SCCP, but if it does work then you could use the console phone built into asterisk rather than IP plumb it into a soft-phone... So asterisk is essentially acting as an SCCP soft-phone itself. No GUI though, but if you're happy typing commands... :) Gordon ___ -- 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] Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
Hi Shocky. It is possible. You should use SIP trunk in CCM and configure some prefix to point to the Asterisk Box. On the Asterisk BOX use SIP peer configuration to make calls trough CCM. You can use some prefix from the both sides and strip it when call arrive at the each side. If you have CCM 5.0 you can use SIP enabled softphone by directly registering info the CCM. Since you have CCM4 you should use SIP trunk to connect CCM and Asterisk. Dimitar Shocky написа: Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. Thanks, Shocky smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
On Friday 10 April 2009 03:33:36 Gordon Henderson wrote: On Fri, 10 Apr 2009, Shocky wrote: Hi, I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? I've never looked at SCCP, but if it does work then you could use the console phone built into asterisk rather than IP plumb it into a soft-phone... So asterisk is essentially acting as an SCCP soft-phone itself. No GUI though, but if you're happy typing commands... :) Gordon That's somewhat encouraging. I'm sure I could get by without a GUI. I guess I need to look in more detail at the state of the SCCP support. Thanks, Shocky -- These are my opinions. Get your own. ___ -- 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] Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
Shocky wrote: This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). It maybe a small nuance, but as a CCM administrator I can understand the refusal to support a roaming H323 or SIP endpoint on CCM. Perhaps if your asterisk box was not mobile, the CCM admins would consider a H323 trunk to your system? I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? The SCCP support in Asterisk is currently limited to asking as a SCCP server, not as an SCCP client. So you cannot use Asterisk to register as a phone to CCM. The SCCP protocol does have a 'trunking' mode, but Cisco barely uses it themselves, and it is geared to low density situation, two-four channels. I am not aware on any effort to duplicate that in chan_skinny. It is conceivable that chan_skinny could be taught to emulate a Cisco endpoint (7965 for example), but the end result would be of limited value. It would have a limited number of lines/channels and the protocol in this use model would not support passing destination information, so it would require a 1-to-1 mapping of a CCM extension to an Asterisk extension. I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? Not quite true. SCCP is a proprietary protocol, but the codecs supported match well with what Asterisk offers, at least the codecs you would likely choose to use. If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. There is a chance, but it depends on working with the CCM admins and how willing they are to create a one-off configuration for you... Dan ___ -- 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] Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
On Friday 10 April 2009 10:53:17 Dan Austin wrote: Shocky wrote: This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). It maybe a small nuance, but as a CCM administrator I can understand the refusal to support a roaming H323 or SIP endpoint on CCM. Perhaps if your asterisk box was not mobile, the CCM admins would consider a H323 trunk to your system? No, I'm not mobile. I telecommute from home. I'm not sure what the reasoning is behind the restriction. Since it's all within the VPN it shouldn't be a security issue. They won't do anything custom for me (they have thousands of users, so probably wouldn't have time). They did say that they are aware of the non-Windows issue, and might eventually provide a solution. I don't know why Cisco won't support Linux, since IP Communicator is written in Java. But nothing I can do about that either. I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? The SCCP support in Asterisk is currently limited to asking as a SCCP server, not as an SCCP client. So you cannot use Asterisk to register as a phone to CCM. The SCCP protocol does have a 'trunking' mode, but Cisco barely uses it themselves, and it is geared to low density situation, two-four channels. I am not aware on any effort to duplicate that in chan_skinny. It is conceivable that chan_skinny could be taught to emulate a Cisco endpoint (7965 for example), but the end result would be of limited value. It would have a limited number of lines/channels and the protocol in this use model would not support passing destination information, so it would require a 1-to-1 mapping of a CCM extension to an Asterisk extension. I only need one line, from my desktop to the CCM server. I'm not sure what might be involved in trying to adapt the chan_skinny code to act as an SCCP client. I've never worked with any VoIP code before. I might be an interesting project to try to merge the chan_skinny code with some SIP client to make an SCCP client. But I'm not sure I'd have time to do it And if I did it on my employer's network, it would end up belonging to them, which would not be a desirable result - if I did it, I would want to release it to the community. Anyone have a CCM server I could legally experiment against without creating code ownership problems? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? Not quite true. SCCP is a proprietary protocol, but the codecs supported match well with what Asterisk offers, at least the codecs you would likely choose to use. Well, that's one bit of good news at least. If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. There is a chance, but it depends on working with the CCM admins and how willing they are to create a one-off configuration for you... That means no chance in my case. Oh well. Dan Thanks for the clarification Dan. Shocky -- These are my opinions. Get your own. ___ -- 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] Can asterisk work with a dynamic IP?
Ronald Wiplinger (Lists) wrote: I know I can setup asterisk without Internet at all and it works as local pbx. Would an asterisk box work with a dynamic IP, with a dyndns name? What must I take care if I try that? I had my * server behind my adsl router that was getting a dynamic Ip address. I simply created a domain for my site at http://www.dyndns.com/ (free) and it worked fine. Al ___ -- 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] Can asterisk work with a dynamic IP?
We're using it here on dynamic IP from our ISP. They provide reverse DNS, which we've simply setup a CNAME to. So, CPE390480Q239432098423.MYISP.COM is cnamed to PBX.MYBUSINESSDOMAIN.COM Did not have to change anything else for this to work. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronald Wiplinger (Lists) Sent: Monday, December 01, 2008 8:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can asterisk work with a dynamic IP? I know I can setup asterisk without Internet at all and it works as local pbx. Would an asterisk box work with a dynamic IP, with a dyndns name? What must I take care if I try that? bye Ronald ___ -- 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] Can asterisk support using different ip for rtp?
I think this is not possible. If you take a look at main/rtp.c there is no config option for an IP address. regards klaus Jun Yin schrieb: some vendors(like alcatel-lucent) developed a kind of sip proxy which includes two parts: one sip signaling module and one or more voice modules. voice modules are responsible for receiving/sending voice traffic(RTP). each voice module has its own IP. so , when the sip signaling part sends out invite packet, it has sip ip in its sip content and different RTP ip in SDP content. (also for 200OK) Now I'm trying to do a test to simulate that product with asterisk. I hope asterisk can sends out different rtp address based on user or domain name. Based on network side, there are many ways to do it: we can configure the network card with multiple IPs, one for SIP and others for RTP. or , we can setup multiple network cards for the asterisk server, one card is for sip signaling and other cards for rtp traffic connecting to different carriers. I think this diagram is reasonable but I was surprised that asterisk does not support it. Maybe asterisk can do this by special configuration? or, there is other free sip proxy software can do this? Thanks. Message: 10 Date: Wed, 25 Jun 2008 05:15:29 -0400 From: Raj Jain [EMAIL PROTECTED] Subject: Re: [asterisk-users] Can asterisk support using different ip for rtp? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin [EMAIL PROTECTED] wrote: Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? SIP is decoupled from RTP, so they can emanate from different IP addresses. Can you present a scenario where this will make sense (in the context where Asterisk is anchoring the media) ? -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk support using different ip for rtp?
25 jun 2008 kl. 03.26 skrev Jun Yin: Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? There's currently no support for that in Asterisk. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk support using different ip for rtp?
On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin [EMAIL PROTECTED] wrote: Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? SIP is decoupled from RTP, so they can emanate from different IP addresses. Can you present a scenario where this will make sense (in the context where Asterisk is anchoring the media) ? -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk support using different ip for rtp?
25 jun 2008 kl. 11.15 skrev Raj Jain: On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin [EMAIL PROTECTED] wrote: Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? SIP is decoupled from RTP, so they can emanate from different IP addresses. Can you present a scenario where this will make sense (in the context where Asterisk is anchoring the media) ? In general, it's quite frequent in larger setups with remote RTP proxys or media servers. However, as I already said, Asterisk can't handle this today. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk support using different ip for rtp?
some vendors(like alcatel-lucent) developed a kind of sip proxy which includes two parts: one sip signaling module and one or more voice modules. voice modules are responsible for receiving/sending voice traffic(RTP). each voice module has its own IP. so , when the sip signaling part sends out invite packet, it has sip ip in its sip content and different RTP ip in SDP content. (also for 200OK) Now I'm trying to do a test to simulate that product with asterisk. I hope asterisk can sends out different rtp address based on user or domain name. Based on network side, there are many ways to do it: we can configure the network card with multiple IPs, one for SIP and others for RTP. or , we can setup multiple network cards for the asterisk server, one card is for sip signaling and other cards for rtp traffic connecting to different carriers. I think this diagram is reasonable but I was surprised that asterisk does not support it. Maybe asterisk can do this by special configuration? or, there is other free sip proxy software can do this? Thanks. Message: 10 Date: Wed, 25 Jun 2008 05:15:29 -0400 From: Raj Jain [EMAIL PROTECTED] Subject: Re: [asterisk-users] Can asterisk support using different ip for rtp? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin [EMAIL PROTECTED] wrote: Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? SIP is decoupled from RTP, so they can emanate from different IP addresses. Can you present a scenario where this will make sense (in the context where Asterisk is anchoring the media) ? -- Raj Jain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk support 20 user's conference?
On Thu, 2008-02-21 at 13:57 +0800, zhao_x_q wrote: HI, Friends, Now I have 20 polycom’s SS2 phones. Can Asterisk support 20 users conference meeting? Yes. And I want to build HD audio conference by using polycom’s 650 ip phone. Can asterisk support G722 HD audio conference? Afaik Asterisk only supports it in 1.6beta. If you need a working solution *now* then have a look at FreeSWITCH which supports wideband and ultra-wideband conferences very well. Regards, Patrick ___ -- 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] Can asterisk support 20 user's conference?
On Fri, Feb 22, 2008 at 03:22:39PM +0100, Patrick wrote: And I want to build HD audio conference by using polycom’s 650 ip phone. Can asterisk support G722 HD audio conference? Afaik Asterisk only supports it in 1.6beta. If you need a working solution *now* then have a look at FreeSWITCH which supports wideband and ultra-wideband conferences very well. Both Asterisk 1.6 and FreeSwitch are not officially a stable release. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Can asterisk support 20 user's conference?
zhao_x_q wrote: HI, Friends, Now I have 20 polycom’s SS2 phones. Can Asterisk support 20 users conference meeting? And I want to build HD audio conference by using polycom’s 650 ip phone. Can asterisk support G722 HD audio conference? Any friend can help me? Thanks Zhao xiaoqiang Whichever codec you use, asterisk needs to be able to transcode to slin so that the channels can be mixed (and back again). Since only Asterisk 1.6.x comes shipped with a G.722 codec, then you are restricted to using this. (there isn't a stable release from 1.6.x yet). Dependant on hardware, asterisk should be able to support 20 users in a conference. I've used Page(phones,d) with lots of phones before, and it seems to work. (creates a dynamic conference with listed phones in it). ___ -- 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] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
Am Dienstag, den 27.11.2007, 18:00 +0100 schrieb Robert Rozman: Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Hi Robert, I researched for something similar about a year ago, and came up with nothing really worth the work. If you can, try to get another ATA that has a real, old-fashioned serial modem plugged into it, and limit that modem to 9600. I think more than that will not work reliably, but you could of course try. The only working implementation of software emulating a modem in conjunction with asterisk I have seen is fax-related, and even there I read from several people that anything better than 9600 is hardly ever achieved. The code there is cranked into fax-use though, not modem use, which would require the PPP bytestream to be off-handed instead of fax parsing. Perhaps iaxmodem would do that No idea. I'd be interested in how you get that working, if you do indeed. BR Anselm ___ --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] Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
Horse hockey... I currently have a *BANK* of PAP2's hooked up to a wide array of analog modems (a USR Total Connect MP/8, two USR Courier V.Everythings and a Digi LANASERVER). After balancing the audio on the pap2's to not feedback audio and reduce chances of echo occurring, I've had no problem maintaining all lines running whether it's within the LAN or from West Coast USA to Europe (fidonet bbs's and x.25 networks) or between the West Coast USA and Australia via SIP point-to-point. The max speed i've obtained is 33.6kbits/s and that's the normal maximum for *non-ISP* configurations. The key things to setup for is: 1.) Steady latency. Latency is the line killer because modems rely on timing. Most of the time (95%) it's not an issue as my routes to the various VSP's I use have a constant strain/timing between myself and them. 2.) Disable Data Compression on the modem and save it in the NVRAM of the modem. (ATK0) Digitized analog signal already has enough lost bits. *DO* however leave Error Correction on. If both modems support it, it helps tremendously even through lag events. 3.) Test, test and retest... Listen to the connection. If it doesn't work at faster speeds, use the ATNx where x is a number from 0 (auto) to 1 (300bps) to 2 (1200bps) etc... so you can figure out the maximum potential of your hardware and voip connections. So yes Virginia, you can do analog modems over VoIP without issue. And pull a decent data rate. All you would need then is to configure the modem and the machine it's connected to as a PPP server then configure the phone to call your modem via *. Anselm Martin Hoffmeister wrote: Am Dienstag, den 27.11.2007, 18:00 +0100 schrieb Robert Rozman: Hi, I have an older phone with touch screen from Philips. It have it connected to Sipura 3000 FXS port and majority of features work ok. But phone also has touchscreen and web browser that I'd love to use for accessing my local web pages. But the phone only allows me to setup ISP phone number (username and password) and it wants to call it to get to Internet. Since it is connected to Sipura3000, call can come to Asterisk and I'd love to somehow fool that device and connect it to local web pages ? I guess I could somehow mimic ISP internet calling feature on local Asterisk server, but have no clue even where to start searching ... Any advice ? Hi Robert, I researched for something similar about a year ago, and came up with nothing really worth the work. If you can, try to get another ATA that has a real, old-fashioned serial modem plugged into it, and limit that modem to 9600. I think more than that will not work reliably, but you could of course try. The only working implementation of software emulating a modem in conjunction with asterisk I have seen is fax-related, and even there I read from several people that anything better than 9600 is hardly ever achieved. The code there is cranked into fax-use though, not modem use, which would require the PPP bytestream to be off-handed instead of fax parsing. Perhaps iaxmodem would do that No idea. I'd be interested in how you get that working, if you do indeed. BR Anselm ___ --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] Can Asterisk match a literal * in extensions.conf
. matches any number of the preceding character, change it to _X.*X. Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal * followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
I don't think * means anything special to A*k, But wouldn't it be: _X.*X. To match as you ask ? (number)(wildcard)*(number)(wildcard) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: 14 September 2007 17:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can Asterisk match a literal * in extensions.conf I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal * followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 11:39:40 Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) The problem you're seeing is that the period is a short-circuit operator. It says if you match everything so far and at least one more character, then you have a match, no need to go any further. You CANNOT match past a '.'. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: . matches any number of the preceding character, change it to _X.*X. That still won't help. Once the Asterisk pattern matching parser sees a period in the pattern, it ignores anything after it. (I'm not exactly happy about that, but that's the way it is.) In short, Asterisk doesn't currently have a good way of handling this situation. Hopefully somebody infinitely smarter than I am will take pity on our plight and give us a some more advanced pattern-matching tools. (Hint, hint) -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
i just met the same problem. i want to match extension that end with a number, but can not find a way. i also found that _.X match all extension, but won't match any caller-id number in dialplan. maybe it is a bug. but it seems not important since _.X is useless anyway. 2007/9/15, Tilghman Lesher [EMAIL PROTECTED]: On Friday 14 September 2007 11:39:40 Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) The problem you're seeing is that the period is a short-circuit operator. It says if you match everything so far and at least one more character, then you have a match, no need to go any further. You CANNOT match past a '.'. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 12:37:11 pm Tilghman Lesher wrote: On Friday 14 September 2007 11:39:40 Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) The problem you're seeing is that the period is a short-circuit operator. It says if you match everything so far and at least one more character, then you have a match, no need to go any further. You CANNOT match past a '.'. Thank you all. I knew I wasn't nuts, but this is the infomation being posted at http://freenum.org/cookbook/ I'll just have to add a prefix. I was hoping to avoid that. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
Anthony Messina wrote: I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work as expected. I would like to be able to do something like: exten = _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal * followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. . must ONLY be the LAST character in a pattern match. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
On 9/14/07, Jared Smith [EMAIL PROTECTED] wrote:
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote:
. matches any number of the preceding character, change it to _X.*X.
That still won't help. Once the Asterisk pattern matching parser sees a
period in the pattern, it ignores anything after it. (I'm not exactly
happy about that, but that's the way it is.) In short, Asterisk doesn't
currently have a good way of handling this situation. Hopefully
somebody infinitely smarter than I am will take pity on our plight and
give us a some more advanced pattern-matching tools. (Hint, hint)
Well, you can have some 10 or so patterns (how long can the number
before be), with X, as X means one digit..
For example:
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
[default-wildcard]
exten = _X.,1,Macro(whatever)
Regards,
Atis
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[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
Jared Smith wrote: On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: . matches any number of the preceding character, change it to _X.*X. That still won't help. Once the Asterisk pattern matching parser sees a period in the pattern, it ignores anything after it. (I'm not exactly happy about that, but that's the way it is.) In short, Asterisk doesn't currently have a good way of handling this situation. Hopefully somebody infinitely smarter than I am will take pity on our plight and give us a some more advanced pattern-matching tools. (Hint, hint) Asterisk's pattern matching is NOT a regex. . means match 1 or more character. It has nothing to do with the preceding characters and must ALWAYS be the last character in a pattern match. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 15:35:47 Anthony Messina wrote:
On Friday 14 September 2007 12:37:11 pm Tilghman Lesher wrote:
On Friday 14 September 2007 11:39:40 Anthony Messina wrote:
I am working on getting freenum.org ISN/ITAD numbers integrated into my
exiting dialplan however I am having trouble getting the extension
matches to work as expected.
I would like to be able to do something like:
exten = _X.*.,1,Macro(isn-outbound...)
The problem you're seeing is that the period is a short-circuit operator.
It says if you match everything so far and at least one more character,
then you have a match, no need to go any further. You CANNOT match past
a '.'.
Thank you all. I knew I wasn't nuts, but this is the infomation being
posted at http://freenum.org/cookbook/
I'll just have to add a prefix. I was hoping to avoid that.
exten = _X.,1,Set(firstpart=${CUT(EXTEN,*,1)})
exten = _X.,n,Set(secondpart=${CUT(EXTEN,*,2)})
exten = _X.,n,GotoIf($[${LEN(${secondpart})}=0]?i,1)
exten = _X.,n,Macro(foo,${firstpart},${secondpart})
--
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Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
On Sat, 2007-09-15 at 00:12 +0300, Atis wrote:
On 9/14/07, Jared Smith [EMAIL PROTECTED] wrote:
On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote:
. matches any number of the preceding character, change it to _X.*X.
That still won't help. Once the Asterisk pattern matching parser sees a
period in the pattern, it ignores anything after it. (I'm not exactly
happy about that, but that's the way it is.) In short, Asterisk doesn't
currently have a good way of handling this situation. Hopefully
somebody infinitely smarter than I am will take pity on our plight and
give us a some more advanced pattern-matching tools. (Hint, hint)
Well, you can have some 10 or so patterns (how long can the number
before be), with X, as X means one digit..
For example:
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _XX*.,1,Goto(default-wildcard|${EXTEN}|1)
Atis--
People are spoiled by regex's, and they want to able to make a match vs.
something I call trailing context. What they don't realize is that
such
matches take (possibly) large amounts of time to complete, because they
loop
or are recursive, depending on the implementation.
Thus, a regex like X+\* (which would mean 1 or more X's followed by
an asterisk. would expand out to the 10 (actually perhaps many more)
lines above-- and run (unexpectedly) slower.
The trouble is, the pattern matcher wouldn't know how long an
expression
like X+\* should be, and could generate hundreds of entries. (if the
pattern
length is limited to 256 chars, say).
It is far better to explode out the entries yourself, as you outlined
above.
You know the max size of incoming stream
murf
[default-wildcard]
exten = _X.,1,Macro(whatever)
Regards,
Atis
--
Steve Murphy
Software Developer
Digium
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Re: [asterisk-users] Can Asterisk match a literal * in extensions.conf
Jared Smith wrote: On Fri, 2007-09-14 at 10:51 -0600, Anthony Francis wrote: . matches any number of the preceding character, change it to _X.*X. That still won't help. Once the Asterisk pattern matching parser sees a period in the pattern, it ignores anything after it. (I'm not exactly happy about that, but that's the way it is.) In short, Asterisk doesn't currently have a good way of handling this situation. Hopefully somebody infinitely smarter than I am will take pity on our plight and give us a some more advanced pattern-matching tools. (Hint, hint) Like PCRE maybe hmm. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk match a literal * in extensions.conf
On Friday 14 September 2007 04:12:48 pm Atis wrote:
exten = _XXX*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _*.,1,Goto(default-wildcard|${EXTEN}|1)
exten = _X*.,1,Goto(default-wildcard|${EXTEN}|1)
excellent sir! thank you! actually, since i'm using this for testing
ISN/ITAD, which currently only has ITAD domains with 3 digits i used:
exten = _XXX*XXX,1,Macro(isn,${EXTEN})
exten = _*XXX,1,Macro(isn,${EXTEN})
exten = _X*XXX,1,Macro(isn,${EXTEN})
(i use the macro to set callerid, etc)
would _XXX*XXX be slower to match than _XXX*. since the . ignores everything
after it as posted by another user?
again, thanks. -a
--
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Re: [asterisk-users] Can asterisk give half-ring periodically for MWI?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. My Uniden phone here uses the stutter dial tone to discover if a message is waiting, and lights up a red light on the phone if there is. How does an analogue phone differentiate between a half ring and a call where someone hangs up quickly? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFG5L2iDQNt8rg0Kp4RApDwAKCakhiLuqAIClqS7M9d7pgq2N0jNQCYuIX0 UwvJNiTkC/544IajMONE+w== =nqBw -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
The SIPuras support it, Asterisk analog does not, as far as I know. Matt wrote: The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
Hi, thanks for the reply. This capability is provided by Sipura ATAs (apparently they do it each time they process SIP REGISTER messages with MWI). The periodic ring works when the same analog phone is connected the Sipura ATA. But not when it is connected to the TDM400p. So to reiterate, what I'm looking for is a way to get the half-ring generated by asterisk and/or TDM400p, WITHOUT the use of a SIP-based ATA. - Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:40:08 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
Do Linksys PAP2Ts support it and if so, where is the setting? On 9/5/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The SIPuras support it, Asterisk analog does not, as far as I know. Matt wrote: The answer, I believe, is yes... but I'm not sure how We had this working on some SPA-2002s from Sipura... but then after an asterisk upgrade it stopped working. I'm not sure if it's a setting in the ATA or asterisk, and we just never needed to pursue it. So the answer is.. yes it can be done.. but unfortunately I'm not sure if it's an asterisk setting or an ATA setting. On 9/5/07, Justin Ridge [EMAIL PROTECTED] wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
On 9/5/07, Matt [EMAIL PROTECTED] wrote: Do Linksys PAP2Ts support it and if so, where is the setting? I don't know about PAP2T but SPA2102 does. Basically anything that is similar to the Sipira-SPA firmware, I don't know how familar you are with them but if your webinterface looks like this: http://www.3cx.com/voip-gateways/images/sipura1.jpg 1) the adapter is based on the original Sipura SPA designs firmwares 2) you should have the option. Honestly I think the PAP2T is one that is based on totally Linksys design. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
That's a clever idea, and it sounds like a viable solution. But (and not knocking your inventiveness in any way), its a bit of a hack to get around what seems like a clear limitation. I'll keep looking for a more elegant solution over the next couple of days, and give this a go if nothing cleaner turns up. Thanks for suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:43:36 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
Yeah, it's a hack for half-rings, but a little less so for putting someone right into voicemailmain without delay. Moj Justin Ridge wrote: That's a clever idea, and it sounds like a viable solution. But (and not knocking your inventiveness in any way), its a bit of a hack to get around what seems like a clear limitation. I'll keep looking for a more elegant solution over the next couple of days, and give this a go if nothing cleaner turns up. Thanks for suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:43:36 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
Agreed. I appreciate your suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 5:55:27 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? Yeah, it's a hack for half-rings, but a little less so for putting someone right into voicemailmain without delay. Moj Justin Ridge wrote: That's a clever idea, and it sounds like a viable solution. But (and not knocking your inventiveness in any way), its a bit of a hack to get around what seems like a clear limitation. I'll keep looking for a more elegant solution over the next couple of days, and give this a go if nothing cleaner turns up. Thanks for suggesting it! - Original Message From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 5, 2007 2:43:36 PM Subject: Re: [asterisk-users] Can asterisk give half-ring periodically for MWI? For my wife I recently set up a cron schedule that, every ten minutes, greps the output of show voicemail users for a new message waiting. Upon finding one, it dumps a call file into asterisk's outgoing directory that rings the house phone and, when one is picked up, it connects the user to voicemailmain. You could put a waittime of just three or four seconds, that should give approx. half a ring and then stop Moj Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can asterisk give half-ring periodically for MWI?
The Polycom hardphones do it by defaultBUT a colleague of mine worked in a large office and she said that monday morning people would be driven mad by almost every phone on the floor making that beeble-bup noise...over and over and over PaulH On Wed, 2007-09-05 at 10:32 -0700, Justin Ridge wrote: Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible Message waiting indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks! Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Can Asterisk hear on two IP addresses? And can I do
Dear Alex; Thanks for your kindly help and answer. The question here is: how asterisk will be able to receive calls at two network cards where each network card has a different IP address. Maybe we need to know if asterisk is doing a hear on the ports only without caring for IP or it is doing a hear only on the IP:port? Any advise? Bilal, There is no technical difference, from Asterisk's point of view, between bridging call legs from two different subnets that have local interfaces versus bridging call legs from two foreign IP destinations. As long as they are routable and reachable, they can be connected. So, I think the short answer to your question is yes, provided I'm understanding it correctly. Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ --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] Can Asterisk hear on two IP addresses? And can I do
Hi Bilal - The question here is: how asterisk will be able to receive calls at two network cards where each network card has a different IP address. Maybe we need to know if asterisk is doing a hear on the ports only without caring for IP or it is doing a hear only on the IP:port? If you look in the sample configuration files, you'll see that iax.conf, sip.conf, mgcp.conf, and skinny.conf all have a line that looks like this: bindaddr= If you set it to an IP address like 192.168.1.150, Asterisk will listen on that address only. If you set it to 0.0.0.0, asterisk will listen on all available ethernet interfaces. You can configure this individually for each different VoIP protocol (sip, iax, mgcp, skinny, etc). So, say you have an asterisk server that has two network cards, one configured to 192.168.1.150 and another configured to 222.6.7.8, and in sip.conf, you set bindaddr=0.0.0.0. In this case, your asterisk server will be listening on 192.168.1.150:5060 and 222.6.7.8:5060. Another sip device could call your asterisk server at either 192.168.1.150 or 222.6.7.8 (provided you don't have any firewalls blocking sip traffic). Does this make sense? - Noah ___ --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] Can Asterisk hear on two IP addresses?
Dear Noah; Thanks a lot. It is the sense :) - Regards Bilal Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --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] Can Asterisk hear on two IP addresses? And can I do
Noah, or anyone actually, question, can the IP address receiving the incoming call be used in extension logic to determine call handling procedures, or maybe a better way to ask is can asterisk provide information as to the IP address on which a request was received? Dave On Mon, 2007-07-23 at 10:10 -0400, Noah Miller wrote: Hi Bilal - The question here is: how asterisk will be able to receive calls at two network cards where each network card has a different IP address. Maybe we need to know if asterisk is doing a hear on the ports only without caring for IP or it is doing a hear only on the IP:port? If you look in the sample configuration files, you'll see that iax.conf, sip.conf, mgcp.conf, and skinny.conf all have a line that looks like this: bindaddr= If you set it to an IP address like 192.168.1.150, Asterisk will listen on that address only. If you set it to 0.0.0.0, asterisk will listen on all available ethernet interfaces. You can configure this individually for each different VoIP protocol (sip, iax, mgcp, skinny, etc). So, say you have an asterisk server that has two network cards, one configured to 192.168.1.150 and another configured to 222.6.7.8, and in sip.conf, you set bindaddr=0.0.0.0. In this case, your asterisk server will be listening on 192.168.1.150:5060 and 222.6.7.8:5060. Another sip device could call your asterisk server at either 192.168.1.150 or 222.6.7.8 (provided you don't have any firewalls blocking sip traffic). Does this make sense? - Noah ___ --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] Can Asterisk hear on two IP addresses? And can I do
Hi Dave - question, can the IP address receiving the incoming call be used in extension logic to determine call handling procedures, or maybe a better way to ask is can asterisk provide information as to the IP address on which a request was received? If you have control (or influence) over the devices calling into your asterisk server, you can always configure a different user to correspond to each ethernet interface. Explicitly configure (or ask) all the devices using Address A to use User A, Address B to use User B, etc. Then you can put the various users in different dialplan contexts, and route calls that way. If you have no control over the incoming calls (i.e. they're all coming in as guests), it may be a tricky thing to implement. - Noah ___ --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] Can Asterisk hear on two IP addresses? And can Ido routing for calls from private to public or public toprivate IP addresses
In general section of sip.conf you can bind sip service to multiple ip addresses. If you setup routing successfully you can send the call received one of ip address through other ip addresses of asterisk. All you have to do is to setup routing the right way. In this conf asterisk can be used both for signaling and media. -Original Message- From: bilal ghayyad [mailto:[EMAIL PROTECTED] Sent: Friday, July 13, 2007 7:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can Asterisk hear on two IP addresses? And can Ido routing for calls from private to public or public toprivate IP addresses Hi List; Can asterisk hear (receive) calls on two IP addresses? How? If yes, then: If I have a VPN router, and my Asterisk server connected to two network cards, one has a private IP address (192.168.0.2) connected to the VPN router (192.168.0.1) and another network card has a private IP address (193.111.196.249) connected directly to the outside default gateway (193.111.196.240), where the VPN default gateway for outside is also (193.111.196.240), then: If I received a call on the network card of IP: 192.168.0.2 then can I route the call for another softswitch server has a public IP address (in another county and another network)? If yes, then is there some condition on this kind of call routing (for example: the communication mode to be full proxy for media and signaling or it can be a proxy only for signaling)? Any help? Regards --- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 0965 9849460 Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --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] Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses
Bilal, There is no technical difference, from Asterisk's point of view, between bridging call legs from two different subnets that have local interfaces versus bridging call legs from two foreign IP destinations. As long as they are routable and reachable, they can be connected. So, I think the short answer to your question is yes, provided I'm understanding it correctly. Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Can Asterisk RAS?
On 6/8/07, Christopher Dobbs [EMAIL PROTECTED] wrote: I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask here before I tried writing my own, just to make sure I wasnot reinventing the wheel. You may want to check out the ZapRAS() dialplan application. I know it's there, and it's supposed to do some sort of RAS stuff, but I've never tried it out. -Jared ___ --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] Can Asterisk RAS?
The IAXMODEM might get you half way there...but if you want to connected it to a windows box (which I assume is why you use the RAS acronym), you'll have to look for remote serial port software. -MD- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Dobbs Sent: Friday, June 08, 2007 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can Asterisk RAS? I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask here before I tried writing my own, just to make sure I wasnot reinventing the wheel. Thank you in advance for any responses. -Chris ___ --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] Can asterisk record the duration of users putting on hold?
Hi, the holdtime in queue log entry is not what we want, that holdtime only records the duration that caller stay in the queue before an agent answers. However what we want is the duration that agent put the customers on hold(i.e music on hold, for SIP, the device will send a re-Invite as I attached last time), actually I already find a way, in sip.conf, there is a option callevents, set to yes, the hold and unhold event will send to manager interface. In version 1.4, manager interface can add a timestamp header for every event, that will help to realize this report feature. Regards, Liangliang On 4/27/07, Humberto Figuera [EMAIL PROTECTED] wrote: Hi Xue Liangliang, If you use queue's then look in queue_log http://www.voip-info.org/wiki/index.php?page=Asterisk+log+queue_log the COMPLETEAGENT and COMPLETECALLER events have this information. COMPLETEAGENT(holdtime|calltime|origposition) The caller was connected to an agent, and the call was terminated normally by the *agent*. The caller's hold time and the length of the call are both recorded. The caller's original position in the queue is recorded in origposition. COMPLETECALLER(holdtime|calltime|origposition) The caller was connected to an agent, and the call was terminated normally by the *caller*. The caller's hold time and the length of the call are both recorded. The caller's original position in the queue is recorded in origposition. -- Humberto Figuera - Using Linux 2.6.20 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --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 -- Regards! Liangliang ___ --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] Can asterisk record the duration of usersputting on hold?
Cross posted from -users to -dev I was looking at adding this functionality in last night. I saw that in app_queue when a call is bridged it determines hold time. Using the following: holdtime = abs((now - qe-start) / 60); and for queue.log the following: (long) (callstart - qe-start) My thoughts were that adding a timer to the hold in res_musiconhold would allow us to calculate hold time while still being channel agnostic. Under the function of moh_alloc() Do something like: chan-holdtimestart = time_t and under moh_release() chan-holdtimeend = time_t chan-holdtimelast = (chan-holdtimeend - chan-holdtimestart) chan-holdtime = chan-holdtime + chan-holdtimelast chan-holdfreq = chan-holdfreq + 1 This would allow for a call to be placed on hold and have that time addeded up as well as keep track of how many time a call was place on hold. It could then be reported as $CDR(callholdtime), this would be separate from the value from app_queue or it could be inherited and then if an agent placed a caller on hold it would add it in to the final number, However being on hold 'waiting' to talk to an agent and being on hold after an agent answers is two different values and should remain as such. Any thoughts -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Humberto Figuera Sent: Thursday, April 26, 2007 3:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can asterisk record the duration of usersputting on hold? Hi Xue Liangliang, If you use queue's then look in queue_log http://www.voip-info.org/wiki/index.php?page=Asterisk+log+queue_log the COMPLETEAGENT and COMPLETECALLER events have this information. COMPLETEAGENT(holdtime|calltime|origposition) The caller was connected to an agent, and the call was terminated normally by the *agent*. The caller's hold time and the length of the call are both recorded. The caller's original position in the queue is recorded in origposition. COMPLETECALLER(holdtime|calltime|origposition) The caller was connected to an agent, and the call was terminated normally by the *caller*. The caller's hold time and the length of the call are both recorded. The caller's original position in the queue is recorded in origposition. -- Humberto Figuera - Using Linux 2.6.20 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --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] Can asterisk record the duration of users putting on hold?
Hi Xue Liangliang, If you use queue's then look in queue_log http://www.voip-info.org/wiki/index.php?page=Asterisk+log+queue_log the COMPLETEAGENT and COMPLETECALLER events have this information. COMPLETEAGENT(holdtime|calltime|origposition) The caller was connected to an agent, and the call was terminated normally by the *agent*. The caller's hold time and the length of the call are both recorded. The caller's original position in the queue is recorded in origposition. COMPLETECALLER(holdtime|calltime|origposition) The caller was connected to an agent, and the call was terminated normally by the *caller*. The caller's hold time and the length of the call are both recorded. The caller's original position in the queue is recorded in origposition. -- Humberto Figuera - Using Linux 2.6.20 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --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] Can Asterisk handle 7000 SIP users?
Copy and paste from my reply to a similar question a couple of weeks ago: 5000 sip registrations is quite a lot, but the more important thing is the number of simultaneous calls. If most of your calls is going to be SIP 2 SIP then I would suggest you use openSER for the SIP registrations and most SIP call routing and use asterisk only for calls to/from PSTN and media such as voicemail and announcements. openSER is made for this and is a lot faster at doing SIP call setup. Then use asterisk where it is good. There are good examples of setting up openSER with asterisk on the net sharing a MySQL DB for users, auth etc. Have a look at http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration and other Asterisk related info at http://openser.org/dokuwiki/doku.php M Dominik Zalewski wrote: Hi All, One of my customer asked me if Asterisk can handle 7000 SIP users. They want anyone that have access to wireless hotspot to make voice calls to the office using software phone or SIP cordless phone. Does anybody did such a setup? What are hardware requirements for server and how much bandwidth I will need using comercial codec? Thank you in advance, Dominik ___ --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] can Asterisk act as a H.323 Gatekeeper?
Pawel wrote: I wonder whether asterisk can play a role of H.323 gatekeeper Not today. Although, disclaimed patches are gladly accepted at http://bugs.digium.com. Jeremy McNamara ___ --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] Can Asterisk Send a TEL URI INVITE?
- Grady Neely [EMAIL PROTECTED] wrote: Can Asterisk emulate this INVITE Configuration? Can it send a tel URI INVITE? No, there is not any support in Asterisk for sending tel: URIs. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] Can Asterisk work in a proxy setting- a challenge
First of all, I assume that since you're asking the question, you want to trunk, or send/receive calls that are on the OTHER SIDE of a proxy from you. Certainly asterisk, as a PBX, can service local IP phones, and connect to PSTN lines, without regard to ANY internet connection. Proxy servers are application based, so the proxy has to understand and process each application specifically. Proxy is generally used to describe a web (http) proxy, but it could be any application (ftp, telnet, etc). There are proxy plug-ins for applications, but those are going to be specific to the application, and the specific proxy program you're using, so the proper thing for you to do would be to first decide what you want to put THROUGH the proxy (H323/SIP/IAX2, etc) and then see if your proxy vendor has a module that'll support that protocol. I don't have any experience trying to do this, but my gut feeling is that it's not going to be feasible - Not because it's technically impossible or anything (although it may be - I'm not an expert in writing proxy plugins), but specifically because there's most certainly going to be some overhead, delay, buffering, etc in the proxy, and it's going to be technically impractical to do high-quality VoIP.. I could be proven wrong though - I think there are H323 modules for some proxy systems. -Steve From: Paul David [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 23, 2006 10:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk work in a proxy setting- a challenge Hi all ! I know asterisk works in a direct internet settings and can also work in a natting settings. But my main question is CAN ASTERISK WORK IN A PROXY SETTING? This is a simple question,but am sure it will challenge the GURU in the house . Prove me wrong ! Expecting your reply in any form . Paul How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call rates. ___ --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] Can Asterisk send RTP to a specific port number?
Jimmy wrote: Can Asterisk send RTP to a specific port number? For instance, I know I can limit INCOMING RTP to certain ports in rtp.conf, but can I limit OUTGOING RTP to a specific port (specifically port 5004 - I'm testing a theory and need to be able to do this) No. SIP/SDP negotiation allows the receiver of the media to dictate where it should be sent; the sender has zero control over 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] can asterisk to say chinese like say english
On Fri, Feb 03, 2006 at 11:32:32PM -0500, Wai Wu wrote: A better solution is write special modules for different language to say 1) a string of digits 2) numbers 3) currencies Translated into Asterisk jargon: patches adding support for Chineese into say.c would be welcomed. Luckily, HEAD seems to contain some support for the language zh in SayUnixTime and SayNumber . 1.2 doesn't, though. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] can asterisk to say chinese like say english
On Sat, 2006-02-04 at 10:59 +0800, Jeffery Chen wrote: this is not just playback recorded voice. this is let asterisk say chinese. how to do this. there have any ideas? its a little more than recorded files for some languages. Say has a few different sub-applications like SayNumber which says a number. Some langauges, like Japanese need to have a context for numbers to say them correctly, so if you record 'ni' for '2' then it will say 'ni' every time it is to say the number two which doesnt work well when there are differences, 2 calls in the queue, 2 minutes left, etc. If this limitation is fine for chineese (I dont know) then you can simply just record a few files, place them into your sounds directory (subdir the language name ie /var/lib/asterisk/sounds/cn) and SetLanguage in your dialplan to 'cn' for example to use the sounds from there. If you dont need it to be so dynamic you can set the language in your config files instead. http://www.voip-info.org/wiki-Asterisk+multi-language has more info -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] can asterisk to say chinese like say english
A better solution is write special modules for different language to say 1) a string of digits 2) numbers 3) currencies -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of trixter aka Bret McDanel Sent: Friday, February 03, 2006 10:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] can asterisk to say chinese like say english On Sat, 2006-02-04 at 10:59 +0800, Jeffery Chen wrote: this is not just playback recorded voice. this is let asterisk say chinese. how to do this. there have any ideas? its a little more than recorded files for some languages. Say has a few different sub-applications like SayNumber which says a number. Some langauges, like Japanese need to have a context for numbers to say them correctly, so if you record 'ni' for '2' then it will say 'ni' every time it is to say the number two which doesnt work well when there are differences, 2 calls in the queue, 2 minutes left, etc. If this limitation is fine for chineese (I dont know) then you can simply just record a few files, place them into your sounds directory (subdir the language name ie /var/lib/asterisk/sounds/cn) and SetLanguage in your dialplan to 'cn' for example to use the sounds from there. If you dont need it to be so dynamic you can set the language in your config files instead. http://www.voip-info.org/wiki-Asterisk+multi-language has more info -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --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] Can Asterisk replace Cisco 5350?
Linuxnizer The Mesmorizer a écrit : Hi, We are using Cisco5350 as a gateway with 2 E1 cards (part# AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my question is can we save some money and use Asterisk + PCI E1 cards? I've had the same issue lately. I need to set up a 4E1 / g.729 solution. Asterisk way - 4 asterisk boxes with 1 E1 card (approx $2k each) - 120 g.729 licences ($1.2k) - 1 SER box to dispatch the calls on the 4 asterisk boxes ($1k) Total: 4 * 2 + 1.2 + 1 = 10.2 In the end, I went on voipsupply.com and saw that they offer Audiocodes mediant 2000 (4 E1 version) for about $10k. Since this box also does SS7 (which is nice if you want to properly interconnect some day), can scale up to 16 E1 and is conveniently packed in a 1U rackable unit, I have decided to go with Audiocodes. Since I am not set up yet, I can't tell wether it is a good decision or not. I will let you know :) Cheers, Jean-Michel. ___ --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] Can Asterisk replace Cisco 5350?
Ignoring SS7, why exactly are you setting up several boxes ? there are quad E1 cards no ? This is way out of my league, but I just want to understand. - Original Message - From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 17, 2005 12:19 Subject: Re: [Asterisk-Users] Can Asterisk replace Cisco 5350? Linuxnizer The Mesmorizer a écrit : Hi, We are using Cisco5350 as a gateway with 2 E1 cards (part# AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my question is can we save some money and use Asterisk + PCI E1 cards? I've had the same issue lately. I need to set up a 4E1 / g.729 solution. Asterisk way - 4 asterisk boxes with 1 E1 card (approx $2k each) - 120 g.729 licences ($1.2k) - 1 SER box to dispatch the calls on the 4 asterisk boxes ($1k) Total: 4 * 2 + 1.2 + 1 = 10.2 In the end, I went on voipsupply.com and saw that they offer Audiocodes mediant 2000 (4 E1 version) for about $10k. Since this box also does SS7 (which is nice if you want to properly interconnect some day), can scale up to 16 E1 and is conveniently packed in a 1U rackable unit, I have decided to go with Audiocodes. Since I am not set up yet, I can't tell wether it is a good decision or not. I will let you know :) Cheers, Jean-Michel. ___ --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] Can Asterisk replace Cisco 5350?
AR Tarzi a écrit : Ignoring SS7, why exactly are you setting up several boxes ? there are quad E1 cards no ? This is way out of my league, but I just want to understand. Because you would need a super monster box to do simultaneous g.729 encoding - and even though I'm not sure it would work properly. Maybe when we have boards which support hardware g.729 encoding this will become a viable option. Cheers, Jean-Michel. ___ --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] Can Asterisk replace Cisco 5350?
From: Jean-Michel Hiver [EMAIL PROTECTED] Linuxnizer The Mesmorizer a écrit : Hi, We are using Cisco5350 as a gateway with 2 E1 cards (part# AS535-DFC-2CE1) to terminate calls. We need 2 more gateways, my question is can we save some money and use Asterisk + PCI E1 cards? I've had the same issue lately. I need to set up a 4E1 / g.729 solution. Asterisk way - 4 asterisk boxes with 1 E1 card (approx $2k each) - 120 g.729 licences ($1.2k) - 1 SER box to dispatch the calls on the 4 asterisk boxes ($1k) Total: 4 * 2 + 1.2 + 1 = 10.2 In the end, I went on voipsupply.com and saw that they offer Audiocodes mediant 2000 (4 E1 version) for about $10k. Since this box also does SS7 (which is nice if you want to properly interconnect some day), can scale up to 16 E1 and is conveniently packed in a 1U rackable unit, I have decided to go with Audiocodes. Since I am not set up yet, I can't tell wether it is a good decision or not. I will let you know :) Cheers, Jean-Michel. Hii Jean-Michel, Couple of notes, I didn't find Audiocodes at voipsupply.com. As far as the E1 is concerned, I think that there are many standards for R2-E1 signaling. Cisco support many variations, not sure if these cards or Asterisk support such wide variaty of R2 signaling. Check Cisco paper on this http://www.cisco.com/en/US/products/sw/iosswrel/ps1826/products_feature_guide09186a00800dc5cf.html Final note, I can get a used Cisco5350 for around $7000 with 2E1 cards, your solution seems a bit pricey compared to a used Cisco. Any advantages or features that come with Asterisk that can't be done with a Cisco5350? Regards, Linuxman. _ Are you using the latest version of MSN Messenger? Download MSN Messenger 7.5 today! http://messenger.msn.co.uk ___ --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] Can Asterisk replace Cisco 5350?
Hii Jean-Michel, Couple of notes, I didn't find Audiocodes at voipsupply.com. This is the product I'm going to order: http://www.voipsupply.com/product_info.php?products_id=213osCsid=8afe5c480fd75d05ce6e5dad5876e3be Final note, I can get a used Cisco5350 for around $7000 with 2E1 cards, your solution seems a bit pricey compared to a used Cisco. Any advantages or features that come with Asterisk that can't be done with a Cisco5350? I don't know Cisco enough to be able to compare. Cheers, Jean-Michel. ___ --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] Can Asterisk accept and relay calls
Title: Can Asterisk accept and relay calls The simple answer is yes, this can be done. Is there anyone in Sydney? I dont know. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Skeeve StevensSent: Thursday, December 08, 2005 8:51 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Can Asterisk accept and relay calls I have a client looking for a cheap solution to relay calls from a remote site to their core voice switching gear. The suggestion has eventuated to Asterisk being the box to accept the calls (from a Voice Carrier) via IP (and have a PRI) and then relay the calls to the central Cisco equipment. I know I'm being a little vague and if there is more information required.. Please let me know. If there are any consultants in Sydney which knows what I'm talking about and can build it please let me know. Skeeve ___ Skeeve Stevens, RHCE Email: [EMAIL PROTECTED] Website: www.skeeve.org - Telephone: (0414) 753 383 Address: P.O Box 1035, Epping, NSW, 1710, Australia eIntellego - [EMAIL PROTECTED] - www.eintellego.net ___ I'm a groove licked love child king of the verse Si vis pacem, para bellum ___ --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] Can Asterisk accept and relay calls
Title: Can Asterisk accept and relay calls ACCA are in Sydney - if you need more info contact me off the list. PaulH - Original Message - From: Kerry Garrison To: [EMAIL PROTECTED] ; 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, December 09, 2005 4:01 PM Subject: RE: [Asterisk-Users] Can Asterisk accept and relay calls The simple answer is yes, this can be done. Is there anyone in Sydney? I dont know. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Skeeve StevensSent: Thursday, December 08, 2005 8:51 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Can Asterisk accept and relay calls I have a client looking for a cheap solution to relay calls from a remote site to their core voice switching gear. The suggestion has eventuated to Asterisk being the box to accept the calls (from a Voice Carrier) via IP (and have a PRI) and then relay the calls to the central Cisco equipment. I know I'm being a little vague and if there is more information required.. Please let me know. If there are any consultants in Sydney which knows what I'm talking about and can build it please let me know. Skeeve ___ Skeeve Stevens, RHCE Email: [EMAIL PROTECTED] Website: www.skeeve.org - Telephone: (0414) 753 383 Address: P.O Box 1035, Epping, NSW, 1710, Australia eIntellego - [EMAIL PROTECTED] - www.eintellego.net ___ I'm a groove licked love child king of the verse Si vis pacem, para bellum ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Can Asterisk act as a media gateway?
Ken D'Ambrosio wrote: I've got an account that's looking at doing some cable/VoIP integration. They were wondering if it were possible to set up something like this: PSTN (T1) - Asterisk - (some VoIP protocol, probably SIP) - Siemens soft switch - their product It sure sounds nice in theory, but I've never tried anything like this. Is there any chance it would work? Yep, we've done ISDN2e -- Asterisk - H.323 - Cisco Call Manager Analogue - Sipura SPA-3000 - which worked really well. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Can Asterisk do This?
Asterisk can authenticate by CLID - it's not a good idea, though as CLID can be spoofed - Original Message - From: Goran Donev To: asterisk-users@lists.digium.com Sent: Thursday, December 01, 2005 10:36 PM Subject: [Asterisk-Users] Can Asterisk do This? I have a client who is looking for the proposed solution and was wondering if any asterisk professionals know if this can be done by asterisk. Calling card platform. Users calling in through local access numbers, they dial local access numbers and make calls through the system to make affordable long distance lines. The lines would be coming to a PRI gateway probably MediaTrix or asterisks directly via a PRI card. They want the calling card platform to identify the users pin through Caller ID. Either if they call from home or they call phone. If they call from a 3rd party location to give them choice to enter their pin to be authorized by the system for them to make a outbound calling. These calls would be registered to their account and would be bill accordingly to the rates given to them. They want easy administration of this software, I saw A2Billing but I didnt see a part to identify the Pin through caller id. They want this software to be GUI driven and to be easy to administer. 2nd part they want is a VOIP Platform for VOIP ATAs for internet clients. They want to be able to attach ATA clients with DID numbers to they can make calls from their homes and receive incoming calls through this system. This part I know Asterisk can do, but I want to know if this is possible with the system they are looking to implement to have the complete package. They want the system to have a nice GUI like AMP to make the changes. If anyone knows how this can be done affordably with a small startup pilot system. Please let me know if this can be done it would be greatly appreciated. Thanks. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Can Asterisk do This?
Goran, Yes, these are possible. You can roll your own, or use an off the shelf system like our ITSP in a box. See my message of a few minutes ago at: http://lists.digium.com/pipermail/asterisk-users/2005-December/136800.html We'll have calling cards and callerid authentication as you describe by the end of the year. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ Goran Donev wrote: I have a client who is looking for the proposed solution and was wondering if any asterisk professionals know if this can be done by asterisk. Calling card platform. Users calling in through local access numbers, they dial local access numbers and make calls through the system to make affordable long distance lines. The lines would be coming to a PRI gateway probably MediaTrix or asterisks directly via a PRI card. They want the calling card platform to identify the users pin through Caller ID. Either if they call from home or they call phone. If they call from a 3^rd party location to give them choice to enter their pin to be authorized by the system for them to make a outbound calling. These calls would be registered to their account and would be bill accordingly to the rates given to them. They want easy administration of this software, I saw A2Billing but I didn’t see a part to identify the Pin through caller id. They want this software to be GUI driven and to be easy to administer. 2^nd part they want is a VOIP Platform for VOIP ATA’s for internet clients. They want to be able to attach ATA clients with DID numbers to they can make calls from their homes and receive incoming calls through this system. This part I know Asterisk can do, but I want to know if this is possible with the system they are looking to implement to have the complete package. They want the system to have a nice GUI like AMP to make the changes. If anyone knows how this can be done affordably with a small startup pilot system. Please let me know if this can be done it would be greatly appreciated. Thanks. ___ --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] Can Asterisk Set CallerID on Broadvoice?
Rusty Dekema wrote: Hey, Has anybody been able to get Broadvoice to pass the callerid string that Asterisk feeds it to the PSTN? If not, can anyone recommend a provider with a similar pricing structure (monthly fee for more-or-less unlimited termination to USA and 20-30 other countries) that will pass callerid (preferably IAX but SIP is fine)? I have found a provider that can provide reliable DIDs and reliable DTMF detection via rfc2833, but I am not that happy with their outgoing pricing structure. So, I would like to have this carrier originate all calls to my DIDs, while sending calls from within my system out via whatever SIP or IAX trunk I choose, using one of my DID numbers as the callerid value. Thanks, Rusty Rusty, Use a few of the paid IAX termination providers so you have failover if one is down. That way you can set caller ID and not worry about your unlimited account being closed or extra charges being billed. Retail accounts like broadvoice offers come with a few gotchas: 1) Business use of residential account is prohibited 2) No resale. If you have customers they need to put the account in their name and pay directly. 3) Opening multiple channels(typically more than 2) at a time will result in extra charges. In many cases that might be 3.9c/minute billed in full minutes and rounded up to nearest cent. So if you have 2 channels in use and an incoming caller simply hangs up when he hears the first auto-attendant prompt, it costs you 4c. I have accounts with 4 IAX termination providers that accept paypal. Some of them offer 0.25 credit so you can test. I started each with a $5 prepay via paypal. So $20 is not much to spend in order to have 4 different termination providers. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk Set CallerID on Broadvoice?
Rusty Dekema wrote: Hey, Has anybody been able to get Broadvoice to pass the callerid string that Asterisk feeds it to the PSTN? If not, can anyone recommend a provider with a similar pricing structure (monthly fee for more-or-less unlimited termination to USA and 20-30 other countries) that will pass callerid (preferably IAX but SIP is fine)? I have found a provider that can provide reliable DIDs and reliable DTMF detection via rfc2833, but I am not that happy with their outgoing pricing structure. So, I would like to have this carrier originate all calls to my DIDs, while sending calls from within my system out via whatever SIP or IAX trunk I choose, using one of my DID numbers as the callerid value. Thanks, Rusty To answer your question about passing caller ID via broadvoice: It is not permitted to change the CID number. It might pass the name when calling another BV account. I have a vonage softphone hooked into asterisk. I can not change caller ID number even when calling other vonage numbers but changing the name seems to work. I called my brother's vonage phone. He got my softphone number on his display but the name he got was Nasty Ho Hotline :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk Set CallerID on Broadvoice?
On Sun, 2005-11-20 at 09:28 -0500, Paul wrote: To answer your question about passing caller ID via broadvoice: It is not permitted to change the CID number. It might pass the name when calling another BV account. Its not permitted becuase broadvoice auths against the caller id data.. To see something funny if you have 2 broadvoice accounts (do not use someone elses that is a bad thing) set account 1 to use the callerid of account 2, you will see a failed invite (at least with 1.0.x) on account 2 for the call. I almost wonder if all SIP devices are smart enough to reject such calls or if they are gonna have problems with people abusing that knowledge to force people to call premium numbers and such. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk Set CallerID on Broadvoice?
trixter aka Bret McDanel wrote: On Sun, 2005-11-20 at 09:28 -0500, Paul wrote: To answer your question about passing caller ID via broadvoice: It is not permitted to change the CID number. It might pass the name when calling another BV account. Its not permitted becuase broadvoice auths against the caller id data.. Did they always do it that way? Are you saying that both number and name sent must match values shown in the web account portal? To see something funny if you have 2 broadvoice accounts (do not use someone elses that is a bad thing) set account 1 to use the callerid of account 2, you will see a failed invite (at least with 1.0.x) on account 2 for the call. I almost wonder if all SIP devices are smart enough to reject such calls or if they are gonna have problems with people abusing that knowledge to force people to call premium numbers and such. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk Set CallerID on Broadvoice?
Has anybody been able to get Broadvoice to pass the callerid string that Asterisk feeds it to the PSTN? If not, can anyone recommend a provider with a similar pricing structure (monthly fee for more-or-less unlimited termination to USA and 20-30 other countries) that will pass callerid (preferably IAX but SIP is fine)? I have found a provider that can provide reliable DIDs and reliable DTMF detection via rfc2833, but I am not that happy with their outgoing pricing structure. So, I would like to have this carrier originate all calls to my DIDs, while sending calls from within my system out via whatever SIP or IAX trunk I choose, using one of my DID numbers as the callerid value. Thanks, Rusty To answer your question about passing caller ID via broadvoice: It is not permitted to change the CID number. It might pass the name when calling another BV account. I have a vonage softphone hooked into asterisk. I can not change caller ID number even when calling other vonage numbers but changing the name seems to work. I called my brother's vonage phone. He got my softphone number on his display but the name he got was Nasty Ho Hotline :) Just a couple of points of clearification When calling from most voip accounts (regardless of which itsp) to pstn telephone numbers, the calleridnum is _sometimes_ passed to the pstn, and is 100% dependent on the services the itsp has implemented. Some will pass it, others will not. Some itsp's will force the calleridnum to whatever your account is associated with, while others accept the calleridnum via your iax or sip connection to them. Calleridname is most frequently not passed by any itsp to the pstn as the central office that terminates the called number does a database lookup to obtain that name (regardless of what you set your name to within asterisk). Some itsp's do subscribe to the database services and will populate that shared database with your calleridname. Write access to the database is rather expensive (relatively speaking), and most low-end / startup itsp's don't subscribe to that service. Even if the itsp subscribes to the database, their write-access is usually limited to only those telephone numbers they are responsible for. (In other words, they can't submit a calleridname of Joe shit the rag for 312-123-4567 if they are not responsible for that number.) That's why you see some itsp's ask for your calleridname in their signup web pages, submitting that name to the database at signup time. Some itsp's allow you to pass the calleridnum via iax or sip, and those that do, don't bother to check to validate whether the number you passed is valid or not (eg, contained in the shared database). So, you might sign up for a voip account that is assigned a DID number of 312-123-4567, but if you set your outgoing calleridnum to 213-456-1234, that is the number the called individual will see. The name they see will be whatever is listed in the shared database regardless of who is responsible for the number and name. (eg, if 213-456-1234 is a SBC number and SBC populated the database with a name like Joe Blow, that will be the name displayed to the called party when you pass the calleridnum as 213-456-1234 from your voip account.) I don't know of any itsp that will accept calleridname by call as that would imply they are submitting that name to the database on a rather immediate basis. There is usually a cost to the itsp for submitting changes (one way or another), and if they tried to do that on a rather immediate basis for each call, their costs would increase for doing so. Note the above pertains to pstn calls and not to voip-to-voip account calls. The voip-to-voip account calls use a calleridnum and calleridname of whatever the itsp happened to implement, which varies by itsp. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk Set CallerID on Broadvoice?
Its not permitted becuase broadvoice auths against the caller id data.. To see something funny if you have 2 broadvoice accounts (do not use someone elses that is a bad thing) set account 1 to use the callerid of account 2, you will see a failed invite (at least with 1.0.x) on account 2 for the call. I almost wonder if all SIP devices are smart enough to reject such calls or if they are gonna have problems with people abusing that knowledge to force people to call premium numbers and such. BV apparently has decided to avoid those callerid problems by implementing limits on their equipment that basically emulates the US pstn telephony standards. Since we all know BV uses non-asterisk equipment for their primary itsp infrastructure, the limitation might be imposed by their softswitch manufacturer. I have a friend that works in a US central office and he reportedly will change the calleridname in the libd database to God Calling, dial a friend, and then change it back. Raises at least some questions. ;) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk proxy a SIP phone to make it look like a Cisco skinny softphone?
Tom Rymes wrote: Why don't you connect to Cisco via Chan_sccp and use a soft or hardphone to connect to asterisk. Like this: Cisco-(chan_sccp)-asterisk-(SIP)-Your phone Just a thought. Do you mean Cisco as the actual phone - instead of the CallManager? I would need to get the SEP* value/etc for all the phones I wanted to call wouldn't I? I was more hoping for: Cisco Phone - [skinny] - Cisco Call Manager -- [skinny] -- Asterisk -- [sip] -- SIP Phone Where the Cisco CM thought Asterisk was an end-device instead of a trunk. -- Cheers Jason Haar Information Security Manager, Trimble Navigation Ltd. Phone: +64 3 9635 377 Fax: +64 3 9635 417 PGP Fingerprint: 7A2E 0407 C9A6 CAF6 2B9F 8422 C063 5EBB FE1D 66D1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk proxy a SIP phone to make it look like a Cisco skinny softphone?
On Oct 17, 2005, at 6:53 PM, Jason Haar wrote: Tom Rymes wrote: Why don't you connect to Cisco via Chan_sccp and use a soft or hardphone to connect to asterisk. Like this: Cisco-(chan_sccp)-asterisk-(SIP)-Your phone Just a thought. Do you mean Cisco as the actual phone - instead of the CallManager? I would need to get the SEP* value/etc for all the phones I wanted to call wouldn't I? I was more hoping for: Cisco Phone - [skinny] - Cisco Call Manager -- [skinny] -- Asterisk -- [sip] -- SIP Phone Where the Cisco CM thought Asterisk was an end-device instead of a trunk. Sorry for the confusing explanation earlier. Yes, I mean what you meant, but I wasn't assuming you were using a Cisco hardphone. Given the setup where CallManager thinks Asterisk is just another SCCP phone (via chan_sccp), you could then connect to asterisk using any SIP or IAX client, not just a Cisco 79XX SIP phone. Disclaimer: I've never done this, so I don't promise it will work! Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk proxy a SIP phone to make it look like a Cisco skinny softphone?
Why don't you connect to Cisco via Chan_sccp and use a soft or hardphone to connect to asterisk. Like this: Cisco-(chan_sccp)-asterisk-(SIP)-Your phone Just a thought. Tom On Oct 16, 2005, at 6:44 PM, Jason Haar wrote: Hi there We have a Cisco VOIP environment here, with hard and softphones. I have a softphone account/etc, but I'm a Linux user and (as far as I'm aware) there is no Cisco softphone for Linux. However I can run Asterisk. So I was wondering if there is a way to convert a SIP phone transaction into a SKINNY transaction so that the Cisco environment thinks it is a Cisco Softphone? I know you can put a trunk in between Asterisk and Cisco Callmanager - but there's no way I'd get the OK for that at this early stage ;-) Thanks! -- Cheers Jason Haar Information Security Manager, Trimble Navigation Ltd. Phone: +64 3 9635 377 Fax: +64 3 9635 417 PGP Fingerprint: 7A2E 0407 C9A6 CAF6 2B9F 8422 C063 5EBB FE1D 66D1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can asterisk send Remote-Party-ID header ???
Atif Rasheed wrote: Kevin P Fleming once said that a patch will be released very soon to send Remote-Party-ID header from Asterisk. and this was said probably in Feburary. Plans changed :-) is that patch released yet or not ? if some please comment, I will really appriciate No, it is not released, and likely will not be until after Asterisk 1.2 is released. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk ring a specific extension based on the number the outside caller dialed?
On Thu, 2005-07-14 at 13:33 -0700, Jeff Ramsey wrote: If I have six channels of a T1 dedicated to Voice, and have 24 phone numbers in a hunt group so that any of the 24 numbers will ring the next available of the six T1 channels, can Asterisk ring a certain extension when a certain number was dialed? For instance, can I dial xxx-xxx-xxx1 and get extension 1, and then dial xxx-xxx-xxx2 and get extension 2? Direct Dialing is what I am trying to accomplish. But still having an IP phone system with auto answer on nights, voicemail, and all of the other features that Asterisk brings to the table. You're looking for DID service (http://www.voip-info.org/tiki-index.php?page=DID). Contact your T1 provider to set this up. -- Bryce Chidester [EMAIL PROTECTED] Rhino Equipment Corp. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk ring a specific extension based on the number the outside caller dialed?
How about with a voice T1 line? Would this work better with that line than with POTS? On 7/8/05 3:06 PM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Jeff Ramsey wrote: I am thinking of having a pots line with multiple numbers on it, and having Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring another desk if the person called xxx-xxx-xxx2, etc. Can Asterisk do this? Not really with POTS. There is some basic support for Distinctive Ring, see the Zap config file. -- Jeff Ramsey MIS Administrator Tubafor Mill, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk ring a specific extension based on the number the outside caller dialed?
I use the distinctive ring detection for our front door intercom, and I've noticed it's not 100% effective. If this is a business type line, I think I might try to find another solution if it's important that it works 100% of the time. -Mishehu Andrew Kohlsmith wrote: On Friday 08 July 2005 17:01, Jeff Ramsey wrote: I am thinking of having a pots line with multiple numbers on it, and having Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring another desk if the person called xxx-xxx-xxx2, etc. Can Asterisk do this? Asterisk can detect distinctive ringing, so if your telco does it this way and it's in a format Asterisk accepts, then yes. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:42cf027c19787645211667! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk ring a specific extension based on the number the outside caller dialed?
Jeff Ramsey wrote: I am thinking of having a pots line with multiple numbers on it, and having Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring another desk if the person called xxx-xxx-xxx2, etc. Can Asterisk do this? Not really with POTS. There is some basic support for Distinctive Ring, see the Zap config file. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk ring a specific extension based on the number the outside caller dialed?
On Friday 08 July 2005 17:01, Jeff Ramsey wrote: I am thinking of having a pots line with multiple numbers on it, and having Asterisk dial my desk if the outside caller dials xxx-xxx-xxx1, and ring another desk if the person called xxx-xxx-xxx2, etc. Can Asterisk do this? Asterisk can detect distinctive ringing, so if your telco does it this way and it's in a format Asterisk accepts, then yes. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk do the following for me ?
You know, that's exactly what I was looking for since the beginning! Unfortunately I only found one of these items for sale in the US and even then I'm not sure if it will be compatible with the European system! Maybe someone can enlighten me once and for all as far as the differences between North America/Europe in telephony. In any case, I already ordered 2 X100P cards which should be arriving in 1 week 1/2. This Asterisk software looks very promising and I might as well build a small Home Office PBX with different extensions! Another stupid question now: anyone knows who does the voices in all these nice systems ? Like, Welcome to Mycompany, for sales press 1, for support press 2 Thanks! Hello, that is even possible without MODEM hardware. It should work with a simple call forwarder/diverter. It connects to both line ends and works more or less like a analogue 2-port pbx with a fixed programmable forwarding number. Offered e.g. in Germany from AUERSWALD (A-BOX) http://www.auerswald.de/int/products/auerswald_box/box_intro.htm or at EBAY ...like here http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItemitem=6386901484 No modems or VoIP equipment except the ATA is needed at all for this ... regards, Jürgen Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk do the following for me ?
On April 24, 2005 11:58 pm, Lee Howard wrote: Certainly I can understand that Digium doesn't stand to make much money selling X100Ps at $10 each, and I can certainly understand them choosing to not sell them. But, by the same token I cannot understand the community's interest in discouraging other folks from joining the community in the way that economically suits them best. It has absolutely nothing to do with what economically suits them best -- it has everything to do with the fact that when you buy a clone X100P you DO NOT KNOW what you're getting. The chipset may be the same but as you can clearly see from searching this very list, the hybrid circuitry (a crucial crucial part of the design) can be VERY different, and even if the hybrid's fine, there are subtle variations in the chipset that can bite you in the ass. If you're just starting out with Asterisk, buck up and buy what is known to work and what is supported by Digium so that if the excrement DOES hit the air-conditioning you at least know your hardware's not at fault and there's someone who will log on to your system to help you fix it. In fact, Digium doesn't even sell the X100P/X101P anymore because the TDM FXO module has a dynamic impedance hybrid (not automatic, you need to specify which telco standard you're wiring in to) and even a nice simple FIR filter you can tune to help eliminate echo and reduce noise. It's simply a better product. Once you know how things work feel free to buy whatever you want. You'll have the understanding to know where to start troubleshooting if things go wrong and you won't be flooding the list and IRC with various Waah, I gots echo, Waah, I can't gets me CID, Waah, Asterisk sucks messages. Unless you know what you're doing (or are personally working with someone who does), buying the Digium stuff *IS* the most economical route. You may get lucky but generally speaking you'll waste far more time and resources pissing about getting the clone card to work than you will if using something known to work. This is along the exact same lines as those who come in here and post I juxt heard abouts this Aestrix thing... whutz teh ABSOLUTELY BARE MINIMUM hardware I need to make this work?! Early optimization (monteary, hardware or even software) is teh suck. It was the fall of the Roman empire, and it'll be the fall of your Asterisk empire if you're not careful. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk do the following for me ?
On April 25, 2005 12:25 am, Kerry Garrison wrote: What year is this? 2005 right? Doesn't everyone on the planet know that you get what you pay for these days? If you want to experiment with Asterisk there is nothing wrong with using clone X100P cards at $6.95 a pop. If you No there is something very wrong with experimenting with Asterisk with a $7 clone card. When it doesn't work the lists get flamed, Asterisk gets blamed, and the experimenter leaves with a bad taste in his mouth about the whole VOIP process. If you're new to Asterisk, use Digium hardware. Once you understand what's going on, buy whatever cheapass shit you can find, at least you'll KNOW that the system does work with the right hardware. fork over some cash for a quality piece of equipment. If you are really diving into Asterisk, you would probably want to get the developer's kit just so you are working with equipment that you will most likely be using in a production environment. For us, our demo systems and backup systems run clone cards but our production systems all use Digium cards. You've got it completely and utterly backward. Until you know what you're doing you have no idea whether the problem is with the card, with Asterisk, with your system or with your configuration. By using known good cards you eliminate two of those potential sources, *AND* you get Digium's technical support department to help with the rest. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk do the following for me ?
The short answer is Yes. However, you would need X100P cards and not regular modem cards. These cards can be found on eBay for about $7 each. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 12:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
