RE: [asterisk-users] Transfer via CTI
Any ideas on this? Thank you. Phil Menico | Chief Technology Officer | 212-951-7632 XTEND Communications | 171 Madison Avenue, New York, NY 10016 | www.xtend.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Menico Sent: Tuesday, April 17, 2007 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Transfer via CTI I used autodial to allow a user to make a call by clicking on a web directory and placing a call file into the Asterisk outgoing directory. That works perfectly for me. What if I want to click on the web directory and transfer my existing call? Is there a comparable interface? Thank you. Phil New York ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer via CTI
I used autodial to allow a user to make a call by clicking on a web directory and placing a call file into the Asterisk outgoing directory. That works perfectly for me. What if I want to click on the web directory and transfer my existing call? Is there a comparable interface? Thank you. Phil New York ___ --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] Asterisk Auto-dial out
Perfect! Thanks a lot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Wednesday, March 07, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk Auto-dial out I am using the * auto-dial out feature but don't want to have to specify a channel (Zap/G2/) to connect to the extension. Current file I use: Channel: Zap/G2/12127778866 # I have to specify a specific channel MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Is there a way that I can just put in the number and have the system decide the channel to use for calling it? What I would like to do: Channel: #=== This number could be # 7645 in which case go via SIP/7645 # 68001 which should go to CiscoSIP/68001 # 12127778866 which would go via Zap/G2/12127778866 MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Based on dialing plan the system should be able to route the call to whatever channel supports dialing that number. You probably want to use the Local channel. Definitely hit the wiki and check it out: http://www.voip-info.org/wiki/view/Asterisk+local+channels The idea behind the local channel is that you can, in effect, drop a call right into a specific part of the dialplan. From there, your dialplan can handle the logic of figuring out which technology and channel to use. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Auto-dial out
I am using the * auto-dial out feature but don't want to have to specify a channel (Zap/G2/) to connect to the extension. Current file I use: Channel: Zap/G2/12127778866 # I have to specify a specific channel MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Is there a way that I can just put in the number and have the system decide the channel to use for calling it? What I would like to do: Channel: #=== This number could be # 7645 in which case go via SIP/7645 # 68001 which should go to CiscoSIP/68001 # 12127778866 which would go via Zap/G2/12127778866 MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632 Priority: 1 Based on dialing plan the system should be able to route the call to whatever channel supports dialing that number. Thank you. Phil Menico ___ --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] Polycom microbrowser issue Error HTTP 406 withIIS
Title: Message Thanks, but we have reasons to want to make it work with IIS. Anyone have a hint of what is the issue? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Thursday, August 24, 2006 6:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS We had a similar problem. Eventuallywe gave up and just used apache. We found that _exactly_ the same content would not work with IIS, but WOULD work with Apache. -Original Message-From: Phil Menico [mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS I have no where else to turn to so if anyone has an answer please send it. I am running sip version 1.6.on a Polycom 601on Asterisk and am unable to get the microbroser to work. The phone returns a 406 error for both idle and services. I can see the file being requested and the subsequent 406 error in the IIS log files. Any ideas on what permissions are needed in IIS or how to format the webpage file? I tried both these 2 files with no luck XHTML file 1: html head /head body Hello phil post /body/html XHTML file 2: ?xml version="1.0" encoding="UTF-8"?html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en" head titleVirtual Library/title /head body PHello phil/P /body/html Log info from IIS: 2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/ - 302 0 295 202 0 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - -2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - http://10.0.1.210:81/Polycom Thank you. Phil ___ --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] [RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS
Title: Message I found this solution from the web and figured I'd share it because it affects all phones getting input from IIS. Map .gif, .jpg, .css etc (in my case I used .xhtml for the Polycom 601) in IIS under your sites: Properties -Virtual directory tab- Configuration - Application configuration - Mappings tab. Make ASP DLL [..\inetsrv\asp.dll] to handle these files. This allows the file with extension XHTML to be passed to the phone and not return a HTTP 406 error (File type not supported by your browser). Hope is helps others. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil MenicoSent: Friday, August 25, 2006 8:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS Thanks, but we have reasons to want to make it work with IIS. Anyone have a hint of what is the issue? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Thursday, August 24, 2006 6:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 withIIS We had a similar problem. Eventuallywe gave up and just used apache. We found that _exactly_ the same content would not work with IIS, but WOULD work with Apache. -Original Message-From: Phil Menico [mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS I have no where else to turn to so if anyone has an answer please send it. I am running sip version 1.6.on a Polycom 601on Asterisk and am unable to get the microbroser to work. The phone returns a 406 error for both idle and services. I can see the file being requested and the subsequent 406 error in the IIS log files. Any ideas on what permissions are needed in IIS or how to format the webpage file? I tried both these 2 files with no luck XHTML file 1: html head /head body Hello phil post /body/html XHTML file 2: ?xml version="1.0" encoding="UTF-8"?html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en" head titleVirtual Library/title /head body PHello phil/P /body/html Log info from IIS: 2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/ - 302 0 295 202 0 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - -2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - http://10.0.1.210:81/Polycom Thank you. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom microbrowser issue Error HTTP 406 with IIS
Title: Message I have no where else to turn to so if anyone has an answer please send it. I am running sip version 1.6.on a Polycom 601on Asterisk and am unable to get the microbroser to work. The phone returns a 406 error for both idle and services. I can see the file being requested and the subsequent 406 error in the IIS log files. Any ideas on what permissions are needed in IIS or how to format the webpage file? I tried both these 2 files with no luck XHTML file 1: html head /head body Hello phil post /body/html XHTML file 2: ?xml version="1.0" encoding="UTF-8"?html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en" head titleVirtual Library/title /head body PHello phil/P /body/html Log info from IIS: 2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/ - 302 0 295 202 0 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - -2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET /Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPointIP-SPIP_601)+libcurl/7.12.1 - http://10.0.1.210:81/Polycom Thank you. Phil ___ --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] WiFi VoIP Handsets..
I use the Zyxel wireless SIP phones with Aerospace (now Cisco) APs and they work just great. My battery is good for all day. Charge them at night. Expensive though. Got them registering to Asterisk, calling Cisco phones via SIP trunks and they sound great. Thank you. Phil Menico 171 Madison Avenue, New York, NY 10016 www.xtend.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Tuesday, May 16, 2006 5:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi VoIP Handsets.. WipeOut wrote: Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? The only phone Senao I tested has a problem when roaming between APs. There's a very high chance of a call being dropped when crossing from 1 AP to another. Do they support things like WPA etc?? not that I know of. Some current phones have been reported to suffer call quality issues just by turning on basic WEP. I have heard the battery life can be a problem.. Is this the case? about 1+ hrs talk time for the Senao. Standby is only 1 day+. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0
Title: Message I have a conflict problem with the eth0 card and wct2xxp digium board. The PRI can receive calls but my network connection is gone. When I "cat /proc/interrupts" I get the following: 1 .. 1 .. .. .. .. 169 0 IO-APIC-level wct2xxp, eth0 .. etc. even before I "modprobe wct2xxp" After I "modprobe wct2xxp" and "modprobe wctdm" and again run "cat /proc/interrupts" I then get: .. .. .. .. .. 169 118489 IO-APIC-level wct2xxp, eth0 201 118497 IO-APIC-level wctdm .. etc How can I force the wct2xxp to load on a separate IRQ? I tried moving the eth0 to IRQ 10 but could not. Any ideas? Thank you. Phil Menico XTEND Communications 171 Madison Avenue, New York, NY 10016 212-951-7632 (Office) 212-951-7683 (Fax) www.xtend.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP hardphones with xml/html/xhtml/microbrowsersupport?
Dan - try Avaya 4610/4620 (WML) and Alcatel iptouch (XML) as well. Phil Menico XTEND Communications -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, January 16, 2006 7:55 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] SIP hardphones with xml/html/xhtml/microbrowsersupport? What hardphones support xml/html/xhmtl/microbrowser? I need an inexpensive SIP hardphone that can run simple applications (queue status, etc). The phones I know of: Aastra 480i, 9112i, 9133i (though limited by 3 LCD lines on the 91xx seems kind of silly) Cisco 79xx Mitel 5235 Polycom IP601 Any others? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Setup Documentation
Title: Message Hello all: Can anyone help mewith finding the best locations for getting setup and other documentation for *. Thank you. Phil Menico www.xtend.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] 911 and lawsuits
Hi Tony, Only used IOWA because I am all the way in NYC. It is very central and advanced as far as I am concerned. Next time I will say NEW JERSEY. Ask the 911 operations center managers about NENA. They should know about it. NENA (National Emergency Number Association) are the ones who try to make policy for E911. Very respected (I believe all volunteers) people in the 911 community with great knowledge of the issues. Thank you. Phil Menico -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Kava Sent: Wednesday, January 07, 2004 5:55 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] 911 and lawsuits Now imagine this person having his SIP phone in IOWA talking to the the telephone switch in New York via VPN and dialing 911. The call will go to NYPD. Why is it the theoretical VoIP user in such examples always seems to be from Iowa or Nebraska? I feel compelled to state that not all people from these states are farmers with pitchforks and SIP phones. I personally live in Omaha, Nebraska, a city I that the Census Bureau indicates I share with 390,006 others, and that may not even include the suburbs. I respectfully request that future examples use other sparsely populated states such as Montana or perhaps a Canadian province from time-to-time. On a more serious note this thread has been very interesting, and this 911 issue will be very important to our organization in the near future. Luckily the 911 operations center is located in our basement so collaboration is easier for us, but when we roll out VoIP we will need to address the redundancy issues and the location issues for remote offices just like any other business. -- Tony Kava Senior Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911 and lawsuits
Terence, Thank you for sharing your thoughts on our judicial system. I am glad you are there and I am here. (i'm not under jurisdiction of a ridiculous judicial system) Anyway, I work in the 911 arena and in the US many states mandate that you have E911 (identify the persons location and call back number to the PSAP) depending on how much space your facility covers. Imagine a company that is in a multi-building campus or multi-floor high rise environment and an employee dials 911 and all the police get is the trunk number at the police station. Imagine if he then faints and cannot tell the PSAP where he is. In classic PBX's the telephone stations are more static and any moves and changes are more hardwired and the changes are sent to the Telco (PS/ALI) database. In VoIP the users are much more mobile. They can pick up their Telephone (VoIP device) go somewhere else and plug into a network jack and call 911. Now imagine this person having his SIP phone in IOWA talking to the the telephone switch in New York via VPN and dialing 911. The call will go to NYPD. There is an organization called NENA that creates guidelines for 911 which most PBX vendors follow. The VoIP issue and 911 is a very big issue and no one has an absolute solution (even though some claim they do). The problem is really discovery of phone devices on the network end points(which end switch and port they are plugged into, gets worse with wireless). The 911 issue is very real for large installations. For smaller ones make sure you put an analog phone at the line coming from the CO or have a single POTS line (usually a FAX line) to use in an emergency. As far as dial tone, yes, even the big PBX's fail but they have 99.999% (at least they claim?) uptime. The cheaper PC you provide the more failures. In one company we put an enhanced 911 system and in the first week a persons life was saved because of it. Don't take 911 lightly, its to save lives not to save law suits. Thank you. Phil Menico -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terence Parker Sent: Tuesday, January 06, 2004 12:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 911 and lawsuits It's just as well that here in Hong Kong employers don't have to worry about being sued by their staff tripping over their own laces ; or microwave oven manufacturers getting sued by old ladies drying off their poodle ; or supermarket owners getting sued by stupid customers who trip over their own kids. In most countries cases such as these would be thrown out the minute they are filed. Of course, these are slight exaggerations insofar as asterisk is concerned - because being able to dial 911 (or 999 as it is in this part of the world) is a much more 'genuine' problem. But nonetheless, it should be the responsibility of the implementor of such a system to ensure that there are adequate measures taken against system failure - such as UPS, or even a primitive analogue phone line somewhere in the home/office. Though I cannot possibly comment regarding 'fear of being prosecuted', simply because I have no reason to fear (i'm not under jurisdiction of a ridiculous judicial system) - I would say that it is a huge shame that a group of people all with the common goal of contributing towards free software projects such as this should even have to worry about things such as lawsuits. If there are people out there who have problems with asterisk, I suggest they just don't use it. To go as far as suing - that is just taking the piss! (sorry, can't think of equivalent non-British term). Terence Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
