Re: [asterisk-users] SDP Issue
:D pretty much true ! On Tue, Jan 24, 2012 at 12:23 PM, Alex Balashov abalas...@evaristesys.comwrote: Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya 4610sw IP Phone
Hi Shaun, So you mean to say that if I will download the SIP firmware the phone will directly work with Asterisk with no Avaya Call Manager (ACM) in between ? My main requirement is to directly make an Avaya phone talk with the asterisk server with no ACM in between. Regards, Aamir Chougule From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ewing Sent: Tuesday, January 24, 2012 3:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Avaya 4610sw IP Phone On 24/01/2012, at 7:46 AM, Jonn Taylor wrote: This phone only works with Avaya IP Ofiice. That's the 5610SW. The 4610SW while sharing the same appearance and also working on the IP Office (with H323 firmware) was designed for the Avaya Communications Manager, and therefore there is SIP firmware available. Aamir, it should work with the SIP firmware. I've registered the 4621SW with Asterisk in the past (the big screen version), and don't remember having any difficulties but this was a few years ago. If the phone is loaded with H323 then you'll need to replace it with SIP before proceeding. -Shaun -- Shaun Ewing | sh...@shaun.netmailto:sh...@shaun.net | http://shaun.net/ Confidentiality Notice: This e-mail, including any attachments, may contain information that is private, confidential, or protected by attorney-client or other privilege. It is intended only for the use of the intended recipient, and is the property of the company originating this e-mail and/or any parent, affiliates and subsidiaries. If you are not the intended recipient, you are hereby notified that any use of the information contained in or transmitted with the communication or dissemination, distribution, or copying of this communication may be prohibited by law. If you have received this communication in error, please immediately return this communication to the sender and delete the original message and any copy of it in your possession. == This message has been scanned by the Tumbleweed MailGate. -- _ -- 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] SDP Issue
LOL :) that really made me chuckle this morning; and very apt for the fact I did not post any fundamental details about the issue. All points duly noted! -- Thanks, Phil - Original Message - Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- 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] SDP Issue
I wasn't so much poking fun at the substance of your post as the fact that you're the only person on this mailing list that posts with a pseudonym, and at that, one evocative of online gaming or forum environments. It just doesn't fit with the culture or the relatively serious, substantive and adult-oriented tenor of this type of list. Do you not notice that? At the risk of being rude, --[ UxBoD ]-- is something that belongs in WoW or a phpBB board full of spotty adolescents. If your real name is Phil, why not post as such? Okay, so maybe you don't want to give out your surname for one reason or another--fair enough. So, post as Phil, or Phil D., if your full name were Phil Deleterious. There's no rule saying you have to. However, the survival of most human social institutions, including those devoted to the exchange of knowledge, is upheld in part by adherence to some conventions of self-presentation and deportment. These conventions help delineate the identity and character of the venue to outsiders, and assist in self-knowledge and affirmation of that character internally. Everyone else here posts with their full name because it communicates: I am a real, adult person solving real-world technical problems related to Asterisk. It is, at least in part, an affirmation of the fact that real personalities--real humans, real identities--underlie participation in Internet forums, especially specialised ones. It is also a nod to the benificent academic origins of the Internet. There are reasons for these conventions. They encapsulate our creation mythos, and they tell us what kind of people we are, as a community. Quite frankly, your From: display name spits on the pedigree, on the storied heritage of how this open-source community came to be. It is not deferential to the accrued wisdom of Internet-focused technical specialists in areas such as Asterisk or IP telephony, and it does not hallow the ground on which we tread. It says that the ROFLcopter has landed!!!111 and lol p0wned teh n00bs. Except, you're being the n00b. Come on, Phil. Self-awareness is important. I know I am being a self-important ass pontificating on this to you. Are you okay with an ASCII art pseudonym that says, I'm a 14 year old playing WoW on a delapidated, slightly yellowed Windows tower draped in dirty underwear? If not for you, why not for us? Please post with a real name. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 6:02 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: LOL :) that really made me chuckle this morning; and very apt for the fact I did not post any fundamental details about the issue. All points duly noted! -- Thanks, Phil - Original Message - Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- 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] SDP Issue
Alex, I would hate for you to have to pen such a long email again using your mobile and do appreciate the comments hence the change to appear more human. -- Thanks, Phil - Original Message - I wasn't so much poking fun at the substance of your post as the fact that you're the only person on this mailing list that posts with a pseudonym, and at that, one evocative of online gaming or forum environments. It just doesn't fit with the culture or the relatively serious, substantive and adult-oriented tenor of this type of list. Do you not notice that? At the risk of being rude, --[ UxBoD ]-- is something that belongs in WoW or a phpBB board full of spotty adolescents. If your real name is Phil, why not post as such? Okay, so maybe you don't want to give out your surname for one reason or another--fair enough. So, post as Phil, or Phil D., if your full name were Phil Deleterious. There's no rule saying you have to. However, the survival of most human social institutions, including those devoted to the exchange of knowledge, is upheld in part by adherence to some conventions of self-presentation and deportment. These conventions help delineate the identity and character of the venue to outsiders, and assist in self-knowledge and affirmation of that character internally. Everyone else here posts with their full name because it communicates: I am a real, adult person solving real-world technical problems related to Asterisk. It is, at least in part, an affirmation of the fact that real personalities--real humans, real identities--underlie participation in Internet forums, especially specialised ones. It is also a nod to the benificent academic origins of the Internet. There are reasons for these conventions. They encapsulate our creation mythos, and they tell us what kind of people we are, as a community. Quite frankly, your From: display name spits on the pedigree, on the storied heritage of how this open-source community came to be. It is not deferential to the accrued wisdom of Internet-focused technical specialists in areas such as Asterisk or IP telephony, and it does not hallow the ground on which we tread. It says that the ROFLcopter has landed!!!111 and lol p0wned teh n00bs. Except, you're being the n00b. Come on, Phil. Self-awareness is important. I know I am being a self-important ass pontificating on this to you. Are you okay with an ASCII art pseudonym that says, I'm a 14 year old playing WoW on a delapidated, slightly yellowed Windows tower draped in dirty underwear? If not for you, why not for us? Please post with a real name. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 6:02 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: LOL :) that really made me chuckle this morning; and very apt for the fact I did not post any fundamental details about the issue. All points duly noted! -- Thanks, Phil - Original Message - Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- 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] SDP Issue
Phil has been using his pseudonym for years, and Alex and his painful/painstaking posting is the only one I have seen even raising the issue. Says even more about Alex than Phil Peg Leg O'Brien Alex Balashov wrote: I wasn't so much poking fun at the substance of your post as the fact that you're the only person on this mailing list that posts with a pseudonym, and at that, one evocative of online gaming or forum environments. It just doesn't fit with the culture or the relatively serious, substantive and adult-oriented tenor of this type of list. Do you not notice that? At the risk of being rude, --[ UxBoD ]-- is something that belongs in WoW or a phpBB board full of spotty adolescents. If your real name is Phil, why not post as such? Okay, so maybe you don't want to give out your surname for one reason or another--fair enough. So, post as Phil, or Phil D., if your full name were Phil Deleterious. There's no rule saying you have to. However, the survival of most human social institutions, including those devoted to the exchange of knowledge, is upheld in part by adherence to some conventions of self-presentation and deportment. These conventions help delineate the identity and character of the venue to outsiders, and assist in self-knowledge and affirmation of that character internally. Everyone else here posts with their full name because it communicates: I am a real, adult person solving real-world technical problems related to Asterisk. It is, at least in part, an affirmation of the fact that real personalities--real humans, real identities--underlie participation in Internet forums, especially specialised ones. It is also a nod to the benificent academic origins of the Internet. There are reasons for these conventions. They encapsulate our creation mythos, and they tell us what kind of people we are, as a community. Quite frankly, your From: display name spits on the pedigree, on the storied heritage of how this open-source community came to be. It is not deferential to the accrued wisdom of Internet-focused technical specialists in areas such as Asterisk or IP telephony, and it does not hallow the ground on which we tread. It says that the ROFLcopter has landed!!!111 and lol p0wned teh n00bs. Except, you're being the n00b. Come on, Phil. Self-awareness is important. I know I am being a self-important ass pontificating on this to you. Are you okay with an ASCII art pseudonym that says, I'm a 14 year old playing WoW on a delapidated, slightly yellowed Windows tower draped in dirty underwear? If not for you, why not for us? Please post with a real name. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 6:02 AM, --[ UxBoD ]--ux...@splatnix.net wrote: LOL :) that really made me chuckle this morning; and very apt for the fact I did not post any fundamental details about the issue. All points duly noted! -- Thanks, Phil - Original Message - Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one of those who rocket-jumps onto the platform and camps with the grenade launcher, trying to stop the reds from capturing the blue flag? I hate how the health and the ammo takes so long to respawn. Is there any way to fix that in deathmatch? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: --[ UxBoD ]-- -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] SDP Issue
On 01/24/2012 07:34 AM, John Novack wrote: Phil has been using his pseudonym for years, and Alex and his painful/painstaking posting is the only one I have seen even raising the issue. Says even more about Alex than Phil Guilty as charged. :-) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] SDP Issue
Doubt really care TBH and the whole reason behind it was a nickname, Unix Bod, given to me by a very knowledgeable friend. I am here to learn from the community and give back where I can. -- Thanks, Phil - Original Message - On 01/24/2012 07:34 AM, John Novack wrote: Phil has been using his pseudonym for years, and Alex and his painful/painstaking posting is the only one I have seen even raising the issue. Says even more about Alex than Phil Guilty as charged. :-) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] SDP Issue
Phil, I applaud both the diplomacy of your responses and your willingness to consider the critique. It was very gentlemanly of you. For the interlopers cashing in cheap shots, my enthusiasm is more restrained. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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
[asterisk-users] ChanSpy : how to know channel name ?
Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? /core show channels verbose/ shows me for example : /SIP/*378680644-2* default SIP/*rs4-2445* sub-uitinternation SIP/*3715320168-2* default SIP*/ibenla2-244* sub-uit789 SIP/*372083610-2* default SIP/*cedhou0-24* sub-uit789 SIP*/travel3-2* pbx-routing SIP/*INTELin-2* pbx-routing SIP/*375382280-2* default SIP/*miq8-2419* sub-uitGSM SIP/*3749378004-* default SIP*/instlpr0-2* sub-uitinternation SIP/*372089170-2* default SIP/*v9q9uLT-* from-GFATRUNK 46 active channels 24 active calls/ If I want to listen to the conversation of /SIP/*miq8-2419*/and /SIP/*375382280-2*/(these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- 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] Pickup calls coming from queues
Great ! I'll test it ASAP and report back here (tomorrow, if possible). 2012/1/23, Alec Davis siva...@paradise.net.nz: How can I test this solution on a 1.8.8.1 system ? If I'm not mistaken, diff https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1. I've just checked out 1.8.8.1 and download my patch from https://reviewboard.asterisk.org/r/1619/diff/raw/ and it applied clean, using the following on a debian lenny box: svn co http://svn.digium.com/svn/asterisk/tags/1.8.8.1 asterisk-1.8.8.1 cd asterisk-1.8.8.1 wget --no-check-certificate https://reviewboard.asterisk.org/r/1619/diff/raw/ mv index.html r1619.diff.txt patch -p0 r1619.diff.txt Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RFE idea for VM application
Hello all, we are using IMAP for the storage of VMs and had a user yesterday his their maxmsg limit (default 100) and was wondering why nobody could leave them messages. I see in /var/log/asterisk/messages that it does write out a warning message of: ast_log(LOG_WARNING, Unable to leave message since we will exceed the maximum number of messages allowed (%u = %u)\n, msgnum, vmu-maxmsg); but I was wondering how feasible it would be to modify the code to add: 1) the mailbox name of the user whom has hit the limit 2) a warning/critical threshold that the user is getting close to the limit using whatever monitoring tool one has available eg. OSSEC the alert could be trapped and the user notified. Is this worthy of an RFE? and possible help from people if I try and create the patch myself ? Thoughts or insults ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] allowguest = yes? no?
Hello I don't understand how I should use the allowguest item: If set to yes, callers from the Net should authenticate, but then, how can I allow strangers to call extensions in my system? allowguest If set to no, this disallows guest SIP connections. The default is to allow guest connections. SIP normally requires authentication, but you can accept calls from users who do not support authentication (i.e., do not have a secret field defined).Certain SIP appliances (such as the Cisco Call Manager v4.1) do not support authentication, so they will not be able to connect if you set allowguest=no: allowguest=no|yes (from Asterisk The future of Telephony) Thank you. -- _ -- 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] RFE idea for VM application
You don’t state which version this is for, but it seems like a simple patch would be for voicemail to play sorry-mailbox-full.wav (standard sound). In lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play the message back since voicemail is one of the larger modules and therefore prone to a better chance of error introduction on an RFE. PSOT – the Unix Bot thing was pretty cool. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws Sent: Tuesday, January 24, 2012 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RFE idea for VM application Hello all, we are using IMAP for the storage of VMs and had a user yesterday his their maxmsg limit (default 100) and was wondering why nobody could leave them messages. I see in /var/log/asterisk/messages that it does write out a warning message of: ast_log(LOG_WARNING, Unable to leave message since we will exceed the maximum number of messages allowed (%u = %u)\n, msgnum, vmu-maxmsg); but I was wondering how feasible it would be to modify the code to add: 1) the mailbox name of the user whom has hit the limit 2) a warning/critical threshold that the user is getting close to the limit using whatever monitoring tool one has available eg. OSSEC the alert could be trapped and the user notified. Is this worthy of an RFE? and possible help from people if I try and create the patch myself ? Thoughts or insults ? -- Thanks, Phil -- _ -- 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] RFE idea for VM application
You don’t state which version this is for, but it seems like a simple patch would be for voicemail to play sorry-mailbox-full.wav (standard sound). In lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play the message back since voicemail is one of the larger modules and therefore prone to a better chance of error introduction on an RFE. PSOT – the Unix Bot thing was pretty cool. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws Sent: Tuesday, January 24, 2012 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RFE idea for VM application Hello all, we are using IMAP for the storage of VMs and had a user yesterday his their maxmsg limit (default 100) and was wondering why nobody could leave them messages. I see in /var/log/asterisk/messages that it does write out a warning message of: ast_log(LOG_WARNING, Unable to leave message since we will exceed the maximum number of messages allowed (%u = %u)\n, msgnum, vmu-maxmsg); but I was wondering how feasible it would be to modify the code to add: 1) the mailbox name of the user whom has hit the limit 2) a warning/critical threshold that the user is getting close to the limit using whatever monitoring tool one has available eg. OSSEC the alert could be trapped and the user notified. Is this worthy of an RFE? and possible help from people if I try and create the patch myself ? Thoughts or insults ? -- Thanks, Phil -- _ -- 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] ChanSpy : how to know channel name ?
Strip off the -x. Just listen to SIP/miq8 and SIP/375382280 in your example. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? core show channels verbose shows me for example : SIP/378680644-2 default SIP/rs4-2445 sub-uitinternation SIP/3715320168-2 default SIP/ibenla2-244 sub-uit789 SIP/372083610-2 default SIP/cedhou0-24 sub-uit789 SIP/travel3-2 pbx-routing SIP/INTELin-2 pbx-routing SIP/375382280-2 default SIP/miq8-2419 sub-uitGSM SIP/3749378004- default SIP/instlpr0-2 sub-uitinternation SIP/372089170-2 default SIP/v9q9uLT- from-GFATRUNK 46 active channels 24 active calls If I want to listen to the conversation of SIP/miq8-2419 and SIP/375382280-2 (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- 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] ChanSpy : how to know channel name ?
Hello, OK thanks. But, I want to listen to the conversation (not just 1 channel out of 2 channels). How then do I use ChanSpy ? On 01/24/2012 03:41 PM, Danny Nicholas wrote: Strip off the --x. Just listen to SIP/miq8 and SIP/375382280 in your example. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, January 24, 2012 7:47 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? /core show channels verbose/ shows me for example : /SIP/*378680644-2* default SIP/*rs4-2445* sub-uitinternation SIP/*3715320168-2* default SIP*/ibenla2-244* sub-uit789 SIP/*372083610-2* default SIP/*cedhou0-24* sub-uit789 SIP*/travel3-2* pbx-routing SIP/*INTELin-2* pbx-routing SIP/*375382280-2* default SIP/*miq8-2419* sub-uitGSM SIP/*3749378004-* default SIP*/instlpr0-2* sub-uitinternation SIP/*372089170-2* default SIP/*v9q9uLT-* from-GFATRUNK 46 active channels 24 active calls/ If I want to listen to the conversation of /SIP/*miq8-2419*/ and /SIP/*375382280-2*/ (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy : how to know channel name ?
I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a bridged call which (it seems to me) should pick up both sides. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, OK thanks. But, I want to listen to the conversation (not just 1 channel out of 2 channels). How then do I use ChanSpy ? On 01/24/2012 03:41 PM, Danny Nicholas wrote: Strip off the -x. Just listen to SIP/miq8 and SIP/375382280 in your example. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? core show channels verbose shows me for example : SIP/378680644-2 default SIP/rs4-2445 sub-uitinternation SIP/3715320168-2 default SIP/ibenla2-244 sub-uit789 SIP/372083610-2 default SIP/cedhou0-24 sub-uit789 SIP/travel3-2 pbx-routing SIP/INTELin-2 pbx-routing SIP/375382280-2 default SIP/miq8-2419 sub-uitGSM SIP/3749378004- default SIP/instlpr0-2 sub-uitinternation SIP/372089170-2 default SIP/v9q9uLT- from-GFATRUNK 46 active channels 24 active calls If I want to listen to the conversation of SIP/miq8-2419 and SIP/375382280-2 (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest = yes? no?
What they are talking about is SIP URI dialling. Let say you have extension 1000 the rings a phone on your system. With allowguest=yes I would be allowed to dial SIP:/1...@yourdomain.com and assuming the context defined in your [General] section had access to exten 1000 I would connect to that phone. With alloweguest=no my call would be rejected. That does not mean that strangers can not call an IVR and get to your 1000 extension or even a DID that point right to it. If you are going to allowguest=yes you need to take carfule note of your contexts so as not to allow strangers access to parts of your dial plan that have, lets say long distance routes. Does that help? Thanks!! Jim On 01/24/2012 09:34 AM, Gilles wrote: Hello I don't understand how I should use the allowguest item: If set to yes, callers from the Net should authenticate, but then, how can I allow strangers to call extensions in my system? allowguest If set to no, this disallows guest SIP connections. The default is to allow guest connections. SIP normally requires authentication, but you can accept calls from users who do not support authentication (i.e., do not have a secret field defined).Certain SIP appliances (such as the Cisco Call Manager v4.1) do not support authentication, so they will not be able to connect if you set allowguest=no: allowguest=no|yes (from Asterisk – The future of Telephony) Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFE idea for VM application
This is in version 1.8 and 10.0 from what I can see. The problem is not that the caller is unaware of the recipients mailbox being full, as they do hear the message, but it is the recipient whom may be completely unaware. If a more verbose warning message was written out we could at least alert the user to the issue. They could then perform some timely and necessary mailbox clean up:) -- Thanks, Phil - Original Message - You don’t state which version this is for, but it seems like a simple patch would be for voicemail to play sorry-mailbox-full.wav (standard sound). In lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play the message back since voicemail is one of the larger modules and therefore prone to a better chance of error introduction on an RFE. PSOT – the Unix Bot thing was pretty cool. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws Sent: Tuesday, January 24, 2012 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RFE idea for VM application Hello all, we are using IMAP for the storage of VMs and had a user yesterday his their maxmsg limit (default 100) and was wondering why nobody could leave them messages. I see in /var/log/asterisk/messages that it does write out a warning message of: ast_log(LOG_WARNING, Unable to leave message since we will exceed the maximum number of messages allowed (%u = %u)\n, msgnum, vmu-maxmsg); but I was wondering how feasible it would be to modify the code to add: 1) the mailbox name of the user whom has hit the limit 2) a warning/critical threshold that the user is getting close to the limit using whatever monitoring tool one has available eg. OSSEC the alert could be trapped and the user notified. Is this worthy of an RFE? and possible help from people if I try and create the patch myself ? Thoughts or insults ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest = yes? no?
On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito asterisk-users-mailing-l...@devito.cc wrote: What they are talking about is SIP URI dialling. Let say you have extension 1000 the rings a phone on your system. With allowguest=yes I would be allowed to dial SIP:/1...@yourdomain.com and assuming the context defined in your [General] section had access to exten 1000 I would connect to that phone. With alloweguest=no my call would be rejected. Thanks for the clarification. Provided I do want strangers to call extensions through an SIP URI instead of using the PSTN, how can I raise security by requiring that they authenticate? Of do you mean that the choice is between - don't allow SIP URI at all (allowguest=no), so strangers can reach extensions only through the PSTN (but it's a waste of money) - allow SIP URI (allowguess=yes) and make sure the context doesn't allow making calls to the PSTN? -- _ -- 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] ChanSpy : how to know channel name ?
Hello, thanks. miq8 is the name of the SIP peer account. So when I know the SIP peer name, and I strip of the numbers of the channel, then I can use ChanSpy. So this answers my original question. The only problem I see : it is Asterisk that gives the channel its name. How do I change this ?? As far as I know, Asterisk randomly gives a channel name which consists of the technology (SIP), the peername (miq8) and some numbers... How to change the channel name ? On 01/24/2012 03:53 PM, Danny Nicholas wrote: I would try chanspy(sip/miq8,b) -- the b flag denotes to only listen to a bridged call which (it seems to me) should pick up both sides. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, January 24, 2012 8:46 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, OK thanks. But, I want to listen to the conversation (not just 1 channel out of 2 channels). How then do I use ChanSpy ? On 01/24/2012 03:41 PM, Danny Nicholas wrote: Strip off the --x. Just listen to SIP/miq8 and SIP/375382280 in your example. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, January 24, 2012 7:47 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? /core show channels verbose/ shows me for example : /SIP/*378680644-2* default SIP/*rs4-2445* sub-uitinternation SIP/*3715320168-2* default SIP*/ibenla2-244* sub-uit789 SIP/*372083610-2* default SIP/*cedhou0-24* sub-uit789 SIP*/travel3-2* pbx-routing SIP/*INTELin-2* pbx-routing SIP/*375382280-2* default SIP/*miq8-2419* sub-uitGSM SIP/*3749378004-* default SIP*/instlpr0-2* sub-uitinternation SIP/*372089170-2* default SIP/*v9q9uLT-* from-GFATRUNK 46 active channels 24 active calls/ If I want to listen to the conversation of /SIP/*miq8-2419*/ and /SIP/*375382280-2*/ (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFE idea for VM application
I would personally rather use a stand-alone daemon to query the mailboxes and send an email to the box owner when he or she reaches a tolerance level rather than depend on an overloaded application that is running God-only-knows what modifications to the original intent (IMAP, Real-Time, Active Directory, any other monkey wrench someone might throw at the original idea of a text file and wav on the actual host). Voicemail.conf presumably has the email of the box owner and the number of messages they can receive. I would suggest adding a warnmsg= to voicemail.conf to work in tandem with maxmsg=. Warnmsg could be a count of messages to send warning at or a percentage of maxmsg. If I ever get a free day, I might try this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws Sent: Tuesday, January 24, 2012 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RFE idea for VM application This is in version 1.8 and 10.0 from what I can see. The problem is not that the caller is unaware of the recipients mailbox being full, as they do hear the message, but it is the recipient whom may be completely unaware. If a more verbose warning message was written out we could at least alert the user to the issue. They could then perform some timely and necessary mailbox clean up:) -- Thanks, Phil _ You don’t state which version this is for, but it seems like a simple patch would be for voicemail to play sorry-mailbox-full.wav (standard sound). In lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play the message back since voicemail is one of the larger modules and therefore prone to a better chance of error introduction on an RFE. PSOT – the Unix Bot thing was pretty cool. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws Sent: Tuesday, January 24, 2012 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RFE idea for VM application Hello all, we are using IMAP for the storage of VMs and had a user yesterday his their maxmsg limit (default 100) and was wondering why nobody could leave them messages. I see in /var/log/asterisk/messages that it does write out a warning message of: ast_log(LOG_WARNING, Unable to leave message since we will exceed the maximum number of messages allowed (%u = %u)\n, msgnum, vmu-maxmsg); but I was wondering how feasible it would be to modify the code to add: 1) the mailbox name of the user whom has hit the limit 2) a warning/critical threshold that the user is getting close to the limit using whatever monitoring tool one has available eg. OSSEC the alert could be trapped and the user notified. Is this worthy of an RFE? and possible help from people if I try and create the patch myself ? Thoughts or insults ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy : how to know channel name ?
It's not random. The Channel Name is Tech/peer-sequence (sequence is in hex). You can control (to a degree) the peer portion in sip.conf/users.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, thanks. miq8 is the name of the SIP peer account. So when I know the SIP peer name, and I strip of the numbers of the channel, then I can use ChanSpy. So this answers my original question. The only problem I see : it is Asterisk that gives the channel its name. How do I change this ?? As far as I know, Asterisk randomly gives a channel name which consists of the technology (SIP), the peername (miq8) and some numbers... How to change the channel name ? On 01/24/2012 03:53 PM, Danny Nicholas wrote: I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a bridged call which (it seems to me) should pick up both sides. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, OK thanks. But, I want to listen to the conversation (not just 1 channel out of 2 channels). How then do I use ChanSpy ? On 01/24/2012 03:41 PM, Danny Nicholas wrote: Strip off the -x. Just listen to SIP/miq8 and SIP/375382280 in your example. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? core show channels verbose shows me for example : SIP/378680644-2 default SIP/rs4-2445 sub-uitinternation SIP/3715320168-2 default SIP/ibenla2-244 sub-uit789 SIP/372083610-2 default SIP/cedhou0-24 sub-uit789 SIP/travel3-2 pbx-routing SIP/INTELin-2 pbx-routing SIP/375382280-2 default SIP/miq8-2419 sub-uitGSM SIP/3749378004- default SIP/instlpr0-2 sub-uitinternation SIP/372089170-2 default SIP/v9q9uLT- from-GFATRUNK 46 active channels 24 active calls If I want to listen to the conversation of SIP/miq8-2419 and SIP/375382280-2 (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] ChanSpy : how to know channel name ?
Of course I can control the name of my SIP-peer. Why do you tell me this ?! Please answer my question : how do I know the channel name so I can ChanSpy the correct channel ? On 01/24/2012 04:13 PM, Danny Nicholas wrote: It's not random. The Channel Name is Tech/peer-sequence (sequence is in hex). You can control (to a degree) the peer portion in sip.conf/users.conf. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, January 24, 2012 9:07 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, thanks. miq8 is the name of the SIP peer account. So when I know the SIP peer name, and I strip of the numbers of the channel, then I can use ChanSpy. So this answers my original question. The only problem I see : it is Asterisk that gives the channel its name. How do I change this ?? As far as I know, Asterisk randomly gives a channel name which consists of the technology (SIP), the peername (miq8) and some numbers... How to change the channel name ? On 01/24/2012 03:53 PM, Danny Nicholas wrote: I would try chanspy(sip/miq8,b) -- the b flag denotes to only listen to a bridged call which (it seems to me) should pick up both sides. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, January 24, 2012 8:46 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, OK thanks. But, I want to listen to the conversation (not just 1 channel out of 2 channels). How then do I use ChanSpy ? On 01/24/2012 03:41 PM, Danny Nicholas wrote: Strip off the --x. Just listen to SIP/miq8 and SIP/375382280 in your example. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, January 24, 2012 7:47 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? /core show channels verbose/ shows me for example : /SIP/*378680644-2* default SIP/*rs4-2445* sub-uitinternation SIP/*3715320168-2* default SIP*/ibenla2-244* sub-uit789 SIP/*372083610-2* default SIP/*cedhou0-24* sub-uit789 SIP*/travel3-2* pbx-routing SIP/*INTELin-2* pbx-routing SIP/*375382280-2* default SIP/*miq8-2419* sub-uitGSM SIP/*3749378004-* default SIP*/instlpr0-2* sub-uitinternation SIP/*372089170-2* default SIP/*v9q9uLT-* from-GFATRUNK 46 active channels 24 active calls/ If I want to listen to the conversation of /SIP/*miq8-2419*/ and /SIP/*375382280-2*/ (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] allowguest = yes? no?
On 01/24/2012 09:03 AM, Gilles wrote: On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito asterisk-users-mailing-l...@devito.cc wrote: What they are talking about is SIP URI dialling. Let say you have extension 1000 the rings a phone on your system. With allowguest=yes I would be allowed to dial SIP:/1...@yourdomain.com and assuming the context defined in your [General] section had access to exten 1000 I would connect to that phone. With alloweguest=no my call would be rejected. Thanks for the clarification. Provided I do want strangers to call extensions through an SIP URI instead of using the PSTN, how can I raise security by requiring that they authenticate? By definition this is impossible. If the caller is a 'stranger', that means you have no knowledge of them prior to their INVITE request arriving at your server. If you have no knowledge of them, then you don't have any 'shared secret', and thus they cannot authenticate to your server. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFE idea for VM application
Hi Danny, Yes I did think about a stand-daemon to do exactly that; and on a side note was even considering using Node.JS :) Though the rationale for considering making the change in app_voicemail.c is that for it to alert on maxmsg it must have already calculated the number of messages in the first place so no real overhead should be introduced. Your suggestion makes perfect sense, and one I had discussed with a colleague, so will have a hack this evening on V10 and see if it would be easy to implement and back port. -- Thanks, Phil - Original Message - I would personally rather use a stand-alone daemon to query the mailboxes and send an email to the box owner when he or she reaches a tolerance level rather than depend on an overloaded application that is running God-only-knows what modifications to the original intent (IMAP, Real-Time, Active Directory, any other monkey wrench someone might throw at the original idea of a text file and wav on the actual host). Voicemail.conf presumably has the email of the box owner and the number of messages they can receive. I would suggest adding a warnmsg= to voicemail.conf to work in tandem with maxmsg=. Warnmsg could be a count of messages to send warning at or a percentage of maxmsg. If I ever get a free day, I might try this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws Sent: Tuesday, January 24, 2012 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RFE idea for VM application This is in version 1.8 and 10.0 from what I can see. The problem is not that the caller is unaware of the recipients mailbox being full, as they do hear the message, but it is the recipient whom may be completely unaware. If a more verbose warning message was written out we could at least alert the user to the issue. They could then perform some timely and necessary mailbox clean up:) -- Thanks, Phil - Original Message - You don’t state which version this is for, but it seems like a simple patch would be for voicemail to play sorry-mailbox-full.wav (standard sound). In lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play the message back since voicemail is one of the larger modules and therefore prone to a better chance of error introduction on an RFE. PSOT – the Unix Bot thing was pretty cool. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Daws Sent: Tuesday, January 24, 2012 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RFE idea for VM application Hello all, we are using IMAP for the storage of VMs and had a user yesterday his their maxmsg limit (default 100) and was wondering why nobody could leave them messages. I see in /var/log/asterisk/messages that it does write out a warning message of: ast_log(LOG_WARNING, Unable to leave message since we will exceed the maximum number of messages allowed (%u = %u)\n, msgnum, vmu-maxmsg); but I was wondering how feasible it would be to modify the code to add: 1) the mailbox name of the user whom has hit the limit 2) a warning/critical threshold that the user is getting close to the limit using whatever monitoring tool one has available eg. OSSEC the alert could be trapped and the user notified. Is this worthy of an RFE? and possible help from people if I try and create the patch myself ? Thoughts or insults ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- 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] ChanSpy : how to know channel name ?
You are either going to be able to listen to SIP/miq8 or you are going to have to know the sequence number like SIP/miq8-1. Maybe you should just use ExtenSpy instead? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Of course I can control the name of my SIP-peer. Why do you tell me this ?! Please answer my question : how do I know the channel name so I can ChanSpy the correct channel ? On 01/24/2012 04:13 PM, Danny Nicholas wrote: It's not random. The Channel Name is Tech/peer-sequence (sequence is in hex). You can control (to a degree) the peer portion in sip.conf/users.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, thanks. miq8 is the name of the SIP peer account. So when I know the SIP peer name, and I strip of the numbers of the channel, then I can use ChanSpy. So this answers my original question. The only problem I see : it is Asterisk that gives the channel its name. How do I change this ?? As far as I know, Asterisk randomly gives a channel name which consists of the technology (SIP), the peername (miq8) and some numbers... How to change the channel name ? On 01/24/2012 03:53 PM, Danny Nicholas wrote: I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a bridged call which (it seems to me) should pick up both sides. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, OK thanks. But, I want to listen to the conversation (not just 1 channel out of 2 channels). How then do I use ChanSpy ? On 01/24/2012 03:41 PM, Danny Nicholas wrote: Strip off the -x. Just listen to SIP/miq8 and SIP/375382280 in your example. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? core show channels verbose shows me for example : SIP/378680644-2 default SIP/rs4-2445 sub-uitinternation SIP/3715320168-2 default SIP/ibenla2-244 sub-uit789 SIP/372083610-2 default SIP/cedhou0-24 sub-uit789 SIP/travel3-2 pbx-routing SIP/INTELin-2 pbx-routing SIP/375382280-2 default SIP/miq8-2419 sub-uitGSM SIP/3749378004- default SIP/instlpr0-2 sub-uitinternation SIP/372089170-2 default SIP/v9q9uLT- from-GFATRUNK 46 active channels 24 active calls If I want to listen to the conversation of SIP/miq8-2419 and SIP/375382280-2 (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] allowguest = yes? no?
On Tue, 24 Jan 2012 09:26:26 -0600, Kevin P. Fleming kpflem...@digium.com wrote: By definition this is impossible. If the caller is a 'stranger', that means you have no knowledge of them prior to their INVITE request arriving at your server. If you have no knowledge of them, then you don't have any 'shared secret', and thus they cannot authenticate to your server. Mmm, so if I want to allow strangers to call us over the Net, I must 1. allowgues=yes 2. make sure the context they enter will not allow them to make calls through the PSTN, either directly (through our plug in the wall) or indirectly (through an ITSP). Thank you. -- _ -- 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] ChanSpy : how to know channel name ?
Extenspy(miq8@default) for miq8. I would either proceed under the assumption that I'm going to be listening to my extensions in the default context or set up an AGI or something to load my needed ext@context information. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, how to use ExtenSpy(extension@context) when conversations are named like this ? : SIP/378680644-2 default SIP/rs4-2445 sub-uitinternation SIP/3715320168-2 default SIP/ibenla2-244 sub-uit789 SIP/372083610-2 default SIP/cedhou0-24 sub-uit789 SIP/travel3-2 pbx-routing SIP/INTELin-2 pbx-routing SIP/375382280-2 default SIP/miq8-2419 sub-uitGSM SIP/3749378004- default SIP/instlpr0-2 sub-uitinternation Can you tell me what is the extension ? How will I know the context ? The context is not always the same... On 01/24/2012 04:32 PM, Danny Nicholas wrote: You are either going to be able to listen to SIP/miq8 or you are going to have to know the sequence number like SIP/miq8-1. Maybe you should just use ExtenSpy instead? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Of course I can control the name of my SIP-peer. Why do you tell me this ?! Please answer my question : how do I know the channel name so I can ChanSpy the correct channel ? On 01/24/2012 04:13 PM, Danny Nicholas wrote: It's not random. The Channel Name is Tech/peer-sequence (sequence is in hex). You can control (to a degree) the peer portion in sip.conf/users.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, thanks. miq8 is the name of the SIP peer account. So when I know the SIP peer name, and I strip of the numbers of the channel, then I can use ChanSpy. So this answers my original question. The only problem I see : it is Asterisk that gives the channel its name. How do I change this ?? As far as I know, Asterisk randomly gives a channel name which consists of the technology (SIP), the peername (miq8) and some numbers... How to change the channel name ? On 01/24/2012 03:53 PM, Danny Nicholas wrote: I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a bridged call which (it seems to me) should pick up both sides. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ChanSpy : how to know channel name ? Hello, OK thanks. But, I want to listen to the conversation (not just 1 channel out of 2 channels). How then do I use ChanSpy ? On 01/24/2012 03:41 PM, Danny Nicholas wrote: Strip off the -x. Just listen to SIP/miq8 and SIP/375382280 in your example. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, January 24, 2012 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ChanSpy : how to know channel name ? Hello list, to use ChanSpy, one needs to know the name of the channel. But on an incoming call from the provider, or an outgoing call to the provider there are always numbers added. How can one then know the channel name ?? core show channels verbose shows me for example : SIP/378680644-2 default SIP/rs4-2445 sub-uitinternation SIP/3715320168-2 default SIP/ibenla2-244 sub-uit789 SIP/372083610-2 default SIP/cedhou0-24 sub-uit789 SIP/travel3-2 pbx-routing SIP/INTELin-2 pbx-routing SIP/375382280-2 default SIP/miq8-2419 sub-uitGSM SIP/3749378004- default SIP/instlpr0-2 sub-uitinternation SIP/372089170-2 default SIP/v9q9uLT- from-GFATRUNK 46 active channels 24 active calls If I want to listen to the conversation of SIP/miq8-2419 and SIP/375382280-2 (these 2 channels have been connected to 1 conversation), how do I use ChanSpy ?? Kind regards; Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join
Re: [asterisk-users] asterisk does not detect menus
I found the fix; in my sip.cfg I changed the following line from DTMF tone.dtmf.level=-15 tone.dtmf.onTime=50 tone.dtmf.offTime=50 tone.dtmf.chassis.masking=0 To DTMF tone.dtmf.level=-9 tone.dtmf.onTime=50 tone.dtmf.offTime=50 tone.dtmf.chassis.masking=0 And it fixed my issue. Thanks, Motty -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, January 23, 2012 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk does not detect menus We had similar problems, updating to the latest 1.8.x seems to have solved the issue for at least one number we were having issues with. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz Sent: Monday, January 23, 2012 2:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk does not detect menus Hello, When I called companies with auto animate menus my system does not seem to detect menus on ther other side. For instance I called this number (407) 886-3338 when I input the ext. number of any option on the list I don't get a response however if I called the same number from my google account or my cell phone number it works fine meaning I can select any option or input ext number. I'm using Asterisk 1.8.4 on Centos 5, Any suggestions are welcome. Thanks, Motty, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1901 / Virus Database: 2109/4761 - Release Date: 01/23/12 -- _ -- 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] Cordless SIP phone
On 1/23/12 4:39 PM, eherr wrote: Where I want to put the new on is outside the range. I thought SIP cordless phones would be better on the range. If you want to extend the range of a DECT basestation you can use repeaters, but you then lose DECT encryption and you can only add up to 6 repeaters around one basestation. Extending your range beyond that requires a proper DECT network and brings you into a whole new cost level. But that can go up to 256x12 handsets and 256x8 (IIRC) simultaneous calls... -- Andreas Sikkema -- _ -- 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] ConfBridge details
On 01/24/2012 04:28 PM, Jeremy Kister wrote:
On 1/23/2012 3:53 PM, Jeremy Kister wrote:
What I'm trying to do is keep track of conferences that are used.
this seems to work:
[macro-confbridge-setup]
exten = s,1,Set(NUM=$[0${NUM} + 1]);
exten = s,n,Set(CONFNO=99${NUM})
exten = s,n,Set(CONFS=${SHELL(asterisk -rx core show channels | awk
'/ConfBridge/ { print $2 }' | awk -F@ '{ print $1 }' | sort | uniq |
grep ${CONFNO} )})
exten = s,n,GotoIf($[${CONFS} = ${CONFNO}]?1)
exten = s,n,Noop(got a new conference# ${CONFNO})
but there's got to be a better way than spawning X shell commands for
'asterisk -rx', right ?
Note that ConfBridge in Asterisk 1.8 is nearly 'experimental', and it
was almost completely rewritten for Asterisk 10. There is gained a lot
more functionality, and much more flexible configuration system, and
dialplan functions that allow access to the sorts of information you are
looking for.
In essence, I would suggest not spending too much time trying to work
the Asterisk 1.8 version of ConfBridge into your dialplan/repertoire,
unless you really need it. The version in Asterisk 10 is much, much better.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
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Re: [asterisk-users] Is there a sip show equivelant.
Is there a way to get a parsable concise feed back from the sip show peers command that is more like the core show channels concise command The issue is the sip show peers uses space delimiter to display the the list but some feilds have values some times and not others. If not what is the exact character count for each column so I can pull back the data that way. Also I have been looking through the code trying to figure out exactly wich fileds are present in the core show channels concise command. It appears that they are different from the core show channels verbose command. What are the exact fields in the core show channels concise and why is there no duration? Thanks Bryant -- _ -- 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] Is there a sip show equivelant.
Question 1 - no
The format is this
#define FORMAT2 %-25.25s %-39.39s %-3.3s %-10.10s %-3.3s %-8s %-11s
%-32.32s %s\n
Question 2
debsphone2*CLI core show channels concise
SIP/1104-051b!default!99!2!Up!Playback!tt-monkeys!1104!!!3!3!(None)!1327
35.1307
debsphone2*CLI core show channels verbose
Channel Context ExtensionPrio State
Application Data CallerIDDuration Accountcode
PeerAccount BridgedTo
SIP/1104-051bdefault 99 2 Up
Playback tt-monkeys110400:00:08
(None)
1 active channel
1 active call
From cli.c (asterisk 10)
e-command = core show channels [concise|verbose|count];
e-usage =
Usage: core show channels
[concise|verbose|count]\n
Lists currently defined channels and some
information about them. If\n
'concise' is specified, the format is
abridged and in a more easily\n
machine parsable format. If 'verbose' is
specified, the output includes\n
more and longer fields. If 'count' is
specified only the channel and call\n
count is output.\n
The 'concise' option is deprecated and will
be removed from future versions\n
of Asterisk.\n;
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a-argc == e-args) {
if (!strcasecmp(a-argv[e-args-1],concise))
concise = 1;
else if (!strcasecmp(a-argv[e-args-1],verbose))
verbose = 1;
else if (!strcasecmp(a-argv[e-args-1],count))
count = 1;
else
return CLI_SHOWUSAGE;
} else if (a-argc != e-args - 1)
return CLI_SHOWUSAGE;
if (!count) {
if (!concise !verbose)
ast_cli(a-fd, FORMAT_STRING2, Channel,
Location, State, Application(Data));
else if (verbose)
ast_cli(a-fd, VERBOSE_FORMAT_STRING2, Channel,
Context, Extension, Priority, State, Application, Data,
CallerID, Duration, Accountcode,
PeerAccount, BridgedTo);
}
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Tuesday, January 24, 2012 4:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is there a sip show equivelant.
Is there a way to get a parsable concise feed back from the sip show peers
command that is more like the core show channels concise command The
issue is the sip show peers uses space delimiter to display the the list but
some feilds have values some times and not others. If not what is the exact
character count for each column so I can pull back the data that way.
Also I have been looking through the code trying to figure out exactly wich
fileds are present in the core show channels concise command. It appears
that they are different from the core show channels verbose command. What
are the exact fields in the core show channels concise and why is there no
duration?
Thanks
Bryant
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Re: [asterisk-users] ConfBridge details
On 1/24/2012 5:32 PM, Kevin P. Fleming wrote: In essence, I would suggest not spending too much time trying to work the Asterisk 1.8 version of ConfBridge into your dialplan/repertoire, unless you really need it. The version in Asterisk 10 is much, much better. good stuff. thanks for the heads-up. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
