Re: [asterisk-users] Caller ID Names
To be sure you could setup a soft phone and see if the caller ID name comes in correctly. On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Hi, In my dialplan I have the following line. same = n,Set(CALLERID(name)=Support) I am expecting this to always set the caller id name to ‘Support’ - however, we are getting calls come in as “Anonymous” with the number as something like “unknown@unknown” We’re using Cisco 7945 phones – I possibly wonder if they are displaying this rather than asterisk not changing it? Anyone had similar experiences before? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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] sip show channelstats reliable?
Thanks but no Adtran here. I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk. From: ewiel...@nyigc.com To: tjrl...@live.com; asterisk-users@lists.digium.com Date: Mon, 19 Jan 2015 13:55:33 -0500 Subject: RE: [asterisk-users] sip show channelstats reliable? I’ve seen something similar with Adtran SIP gateways.When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets.BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is very annoying. At some point I’ll try and arrange a slugfest between Digium and Adtran and they can figure out why it doesn’t work. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd R. Sent: Monday, January 19, 2015 1:45 PM To: Asterisk-Users List Subject: Re: [asterisk-users] sip show channelstats reliable? Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. PeerCall IDDurationRecv: PackLost( %)JitterSend: PackLost(%)Jitterx.x.x.x5531341d06b00:07:42023123063836(73.41%)0.02310200(0.00%)0.0007 Peer IP changed to protect the innocent :-) From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Mon, 19 Jan 2015 12:17:25 -0600 Subject: [asterisk-users] sip show channelstats reliable?I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from. Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues. All I have is the loss that's shown from this command with no real network stats to back it up. Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command? Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info. Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk. The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion. Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL -- _ -- 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] sip show channelstats reliable?
Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter x.x.x.x 5531341d06b 00:07:42 023123 063836 (73.41%) 0. 023102 00 ( 0.00%) 0.0007 Peer IP changed to protect the innocent :-) From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Mon, 19 Jan 2015 12:17:25 -0600 Subject: [asterisk-users] sip show channelstats reliable? I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from. Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues. All I have is the loss that's shown from this command with no real network stats to back it up. Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command? Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info. Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk. The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion. Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL -- _ -- 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] sip show channelstats reliable?
I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the same IP's to/from. Since I don't 100% control the network my gear is on, I need something outside of Asterisk to show the network engineer to convince here and myself that there are network issues. All I have is the loss that's shown from this command with no real network stats to back it up. Is there a magic command in CentOS anyone can recommend to diagnose and match up the issues shown in Asterisk using this command? Moving gear around on the network changes the info Asterisk shows a LOT. For example, if I point traffic to the main physical gateway I get loss to a particular customer's IP (their PBX), if I move it to another place on the network (as a VM) their IP is good and other customers IP's start showing loss using the channelstats info. Driving me freakin' crazy. It does appear there are network issues causing my troubles but I can't get help if I can't point to some hard and fast issues outside of Asterisk. The only thing I have right now is collissions showing on one of a few of our pfSense devices but they are virtual running on XenServer, still this would indicate a problem in my opinion. Thanks in advance for any assistance on this issue. Stepping back from the ledge now LOL -- _ -- 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] ITSP Gateway Solution?
Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP trunks and our customer PBXs. I love and understand Asterisk but the company I am working for is looking for a more Commercial type solution where we can go to a vendor for support etc. I know, we can get Asterisk support etc.. It's not my decision and I sort of get why they are leaning away from Asterisk, I just don't agree. I need to at least explore other options for more appliance products that will do the job the Asterisk boxes are doing now but, with a simple interface to add/remove trunks, DIDs etc. Integrated security and billing options/add-ons would be great. I know Digium offers appliance solutions but they don't seem to be anywhere near the power of what we are currently using. One big advantage I could see is going diskless but, I am really not sure whats out there, I am just kicking tires at the moment. The best of all worlds would be something with commercial support, a good GUI, billing and security built in but all based on the Asterisk core which I can understand :-) Again, just kicking tires as I can't just scream Asterisk and not be willing to look around to see what's out there. Everything I see out there seems to want to Transcode and such.. All we need is something to do SIP to SIP, no TDM here at all. Some codec support beyond G711 of course but that's it. I know there is every reason to do all this with Asterisk and that is my preference but in this case, I have lots of folks that lean more towards commercial products and I have not been able to completely sell them on the joy and flexibility of Asterisk. I don't want a Virtual PBX GUI solution, I want something that is built to be a work-horse, as a gateway only. No extensions, voicemail, ring groups or any of that. Just calls in/out to/from trunks, security and billing. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting Music on Hold with the Manager Interface
Thanks Matt. I tried that already, no luck. Still, I get blank nothingness instead of MOH. I will try again just to be sure I didn't miss something. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Date: Mon, 27 Oct 2014 08:51:42 -0500 Subject: Re: [asterisk-users] Setting Music on Hold with the Manager Interface From: mjor...@digium.com To: tjrl...@live.com; asterisk-users@lists.digium.com On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote: Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this.# Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); Use the CHANNEL function: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL Action: SetvarChannel: (your channel name here)Variable: CHANNEL(musicclass)Value: (your MoH class here) -- Matthew Jordan Digium, Inc. | Engineering Manager445 Jan Davis Drive NW - Huntsville, AL 35806 - USACheck us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting Music on Hold with the Manager Interface
Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this.# Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); -- _ -- 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] Lost audio on forwarded calls
OK, been messing with Asterisk for a long time and I have my opinion on where the issues lies but sometimes it's just nice to see what others think that can relate :-) Here goes.. Inbound calls flow like this:Tier 1 Provider (SIP) Asterisk 1.8 Name Brand PBX - Calls work fine Outbound calls flow like this:Name Brand PBX Asterisk 1.8 Tier 1 provider (SIP) - Calls work fine Problem is being reported on that many (not all) calls have no audio when they are forwarded. Example of forwarded call:Inbound call comes in from Tier 1 Provider Asterisk 1.8 Name Brand PBX Name Brand PBX then forwards the call back out to users cell phone:Name Brand PBX Asterisk 1.8 Tier 1 provider No audio a large percentage of the time. It's my opinion that the Asterisk box only sees the forwarded call as a regular outbound call and forwards it on to the Tier 1 provider then to the users cell phone. I don't see how Asterisk even knows or cares if it was forwarded within the Name Brand PBX. The Name Brand PBX is the one making the connection of the inbound and outbound call. All other inbound and outbound calls are fine, audio is only lost when the Name Brand PBX connects the two calls and creates the forward. Thoughts? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and alternate RTP ports
Been working with Asterisk for a long time but this is the first time I have dealt with this issue. I am setting up an Asterisk box (FreePBX not my choice) to interface with an e911 provider. They say their switches only listen for RTP on ports 2-21001 which is outside the normal range Asterisk listens on 1-2. I wish I knew more about this topic but since I have never had an issue interfacing with providers, ITSP etc., I just haven't had a need to know. I get audio on some calls and others not so much. How do I deal with this? I don't really want to change the RTP ports that Asterisk listens on because this is a production system with trunks pointing to several other providers etc. The 911 provider says I don't need to listen on 2-21001, that's just what they listen on. In fact, they say this exactly You can listen on what you want, as long as the RTP port is sent to us in the INVITE SDP info.. Any assistance with solving this issue would be greatly appreciated, I have done my digging in Google etc before asking here as always. Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API
Search google for Asterisk Manager Interface. For the most part, if you have raw Asterisk installed then that's what you get and have to build what you want on top of it or hire a developer to do it. Date: Fri, 10 Jan 2014 12:12:47 -0500 From: szilvertho...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk API Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen -- _ -- 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] Text to Speech Engine
Luminvox is one.. There are others out there.. Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448 From: jpra...@gmail.com Date: Fri, 10 Jan 2014 12:16:43 -0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Text to Speech Engine Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- 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] Text to Speech Engine
Actually, scratch that.. Luminvox is not text to speech it's speech recognition software. Got this mixed up and turned around :-) Anyhow, see the link I posted earlier, it's got some good info to get you started. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Fri, 10 Jan 2014 14:42:27 -0600 Subject: Re: [asterisk-users] Text to Speech Engine Luminvox is one.. There are others out there.. Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448 From: jpra...@gmail.com Date: Fri, 10 Jan 2014 12:16:43 -0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Text to Speech Engine Hello, Anyone know good quality text to speach engine for building IVRs for asterisk. Open-source will be nice, but I wont mind paying for thing really good. Regards, -Jai -- _ -- 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] Convert Asterisk Appliance (AA50) to Open Asterisk?
May not be what you are looking for exactly but search Google for Nerd Vittles BeagleBone. I am not suggesting you use that exact solution but, reading the article with give you all sorts if ideas about what you could use in your situation. The BeagleBone is a small form factor computer like the Raspberry Pi but more powerful for not much more $$. They have a few builds you can out in it with full instructions. For your analog lines you could just use a couple if cheap SIP gateway devices which I think run $50-80/ea. USD. In a small environment, this is what I would be using these days and I am itching to build something on the BeagleBone boards. Total solution with a few SIP gateway devices should not cost you much at all. The boards are under $50 and a nice little case can be had for under $18 or maybe way less. You could likely use your existing device as a pass through for your analog lines if you don't want to purchase the SIP gateways right away. Let me know if you go down this road and what your results are. Good luck. On Dec 28, 2013, at 12:38 PM, Lincoln King-Cliby linc...@controlworks.com wrote: Hi All, Thanks for all of the help I've been given in the past and info I've picked up from this list over the years. I have an official Asterisk appliance (the AA50) running my PBX at home (we previously also had an AA50 in a satellite office-that one was recently retired and replaced with Asterisk running on commodity server hardware). Anyway - the AA50 software/Asterisk version is beyond outdated at this point, and the GUI has done nothing but infuriate me. Has anyone - or does anyone know if it's possible to - replace the commercial Linux/Asterisk running on the AA50 with another Linux flavor (say Ubuntu) and current open source Asterisk (ideally 11.something with Gareth's Cisco patch). I don't need - or want - a pretty GUI... just something I can SSH into and perhaps manhandle config files with Nano or something similar - worst case, something I can FTP/TFTP configuration files to. If that isn't feasible, anything low power/low profile/low cost that's particularly popular these days [bonus points if it's wall mountable/about the same size as the AA50]? My demands really aren't that severe -- one FXO, two FXS, a SIP trunk to the office (via hardware VPN), and maybe a half dozen Cisco 79xx phones. If it's not already apparent, I'm a relative Linux newb, but I'm farly well versed in patching and building Asterisk from source and generally getting things done once I have the pointers. Thanks in advance -- and happy new year! -- Lincoln King-Cliby, CTS, DMC-D Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Silver) ControlWorks Consulting, LLC Crestron Services Provider http://www.controlworks.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] Answering agent
I do this by writing custom CDR. I write the agents extension write into the CDR records. This makes is easy to just parse through the CDR and get all the info you need about the call. Google something like asterisk custom CDR On Nov 29, 2013, at 11:36 AM, Leandro Dardini ldard...@gmail.com wrote: Hello friends, when a call arrives in the queue, a CDR record is created, but there is no info about which agent has picked up the call. I can find that info only in queue_log. Is there a way to have that info in the CDR or maybe in a variable in the h context, when the call is ended? Leandro -- _ -- 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] Amazon, Asterisk and reliability beyond a hobby system?
Did you have the externalip setting in sip.conf set to the Elastic IP? Date: Sat, 23 Nov 2013 23:42:36 -0500 From: ja...@fivecats.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? On 11/22/2013 12:52 PM, Todd R. wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. I have only one small datapoint. I ran an EC2 microinstance with Asterisk and a dozen offboard users. The only problem I had was SIP wasn't dealing well with the Elastic IP one-to-one NAT that Amazon uses. I had the usual Asterisk/NAT issues of one-way audio. I eventually moved from Amazon to Linode to get away from the NAT issues. Once I did that, everything worked fine, but again it was only a dozen 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] Amazon, Asterisk and reliability beyond a hobby system?
Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Mon, 18 Nov 2013 18:33:38 -0600 Subject: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? Took me a while but I have finally embraced cloud computing and all the benefits. The only thing I have yet to feel comfortable about putting in the cloud is real live Asterisk boxes to be used in production. I know it's being done because as far as I know Twilio is using Amazon for their Asterisk boxes. I have read all the fun articles on building hobby type systems and that's all great. What I really need to hear is from those that have deployed Asterisk in Amazon or Digital Ocean and how many simultaneous calls they are pushing through it and what the call quality and reliability has been. Right now I am still using dedicated hardware but I could become much more redundant and scale much faster using Amazon or Digital Ocean. Thanks in advance for any information from those that have already been down this road... -- _ -- 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] Amazon, Asterisk and reliability beyond a hobby system?
I would have said the same thing a while back but, I can't ignore the fact that there have been what seems to be many Virtualization success stories. The idea that Asterisk just likes to be on it's own dedicated hardware has always caused me to prefer dedicated hardware. But, is the possibility of a single piece of hardware failing better than something that will likely never just flat out die? I know there are high availability solutions out there and it's not that I don't have backups and disaster recovery plans in place. I just want to make things far better regarding redundancy, recovery and scalability and virtualization is hard to beat when you start talking about these things. There are definitely people/companies using virtualized Asterisk solutions successfully, so I feel like it can be done. Asterisk has come a long way since I first starting messing with Asterisk and so has Asterisk itself. So, I am trying to determine what is bad, what to look out for in terms of virtualizing. If it's still as bad of an idea as it was say 5 years ago, then I need to understand why and if there is a work around. At this point, the benefits of virtualizing my Asterisk boxes are too many to count. So, if I can't find any concrete reasons to NOT do this beyond That's a bad idea then I am going to give it a go. If I do, I am looking for any advice good or bad from those that have gone down this road successfully or with miserable failure. My opinion all along has been Asterisk + Virtualization + Real Live Production Use = BAD IDEA! Now, I am trying to figure out if that's just the opinion of an old man (sort of old) who just doesn't want to accept that virtualization if a better way (in terms of Asterisk). So, I am hoping for people to tell me why Amazon AWS specifically is a good or bad idea with as much detail as possible. Thanks! To: tjrl...@live.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? Date: Fri, 22 Nov 2013 13:04:44 -0500 From: cov...@ccs.covici.com I would thinktwice about Amazon -- and virtual in general is not a good idea for this sort of thing. I have seen messages about bad results with amazon specifically. Todd R. tjrl...@live.com wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Mon, 18 Nov 2013 18:33:38 -0600 Subject: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? Took me a while but I have finally embraced cloud computing and all the benefits. The only thing I have yet to feel comfortable about putting in the cloud is real live Asterisk boxes to be used in production. I know it's being done because as far as I know Twilio is using Amazon for their Asterisk boxes. I have read all the fun articles on building hobby type systems and that's all great. What I really need to hear is from those that have deployed Asterisk in Amazon or Digital Ocean and how many simultaneous calls they are pushing through it and what the call quality and reliability has been. Right now I am still using dedicated hardware but I could become much more redundant and scale much faster using Amazon or Digital Ocean. Thanks in advance for any information from those that have already been down this road... -- _ -- 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 Alternatives: -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth
Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
Oh and, I could be wrong but.. I suspect Twilio is one of the companies doing big things with Asterisk on AWS specifically. I am 90% sure at this point that Twilio uses Asterisk as the base for their product. When I emailed them and asked them where their voice gateways were they mentioned something about Amazon's servers which I assumed to mean they were using Amazon's cloud services. The possibility of Twilio pushing tons of calls through virtualized Asterisk boxes is part of what has made me so curious about going down this road again. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Fri, 22 Nov 2013 12:18:35 -0600 Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? I would have said the same thing a while back but, I can't ignore the fact that there have been what seems to be many Virtualization success stories. The idea that Asterisk just likes to be on it's own dedicated hardware has always caused me to prefer dedicated hardware. But, is the possibility of a single piece of hardware failing better than something that will likely never just flat out die? I know there are high availability solutions out there and it's not that I don't have backups and disaster recovery plans in place. I just want to make things far better regarding redundancy, recovery and scalability and virtualization is hard to beat when you start talking about these things. There are definitely people/companies using virtualized Asterisk solutions successfully, so I feel like it can be done. Asterisk has come a long way since I first starting messing with Asterisk and so has Asterisk itself. So, I am trying to determine what is bad, what to look out for in terms of virtualizing. If it's still as bad of an idea as it was say 5 years ago, then I need to understand why and if there is a work around. At this point, the benefits of virtualizing my Asterisk boxes are too many to count. So, if I can't find any concrete reasons to NOT do this beyond That's a bad idea then I am going to give it a go. If I do, I am looking for any advice good or bad from those that have gone down this road successfully or with miserable failure. My opinion all along has been Asterisk + Virtualization + Real Live Production Use = BAD IDEA! Now, I am trying to figure out if that's just the opinion of an old man (sort of old) who just doesn't want to accept that virtualization if a better way (in terms of Asterisk). So, I am hoping for people to tell me why Amazon AWS specifically is a good or bad idea with as much detail as possible. Thanks! To: tjrl...@live.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? Date: Fri, 22 Nov 2013 13:04:44 -0500 From: cov...@ccs.covici.com I would thinktwice about Amazon -- and virtual in general is not a good idea for this sort of thing. I have seen messages about bad results with amazon specifically. Todd R. tjrl...@live.com wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Mon, 18 Nov 2013 18:33:38 -0600 Subject: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system? Took me a while but I have finally embraced cloud computing and all the benefits. The only thing I have yet to feel comfortable about putting in the cloud is real live Asterisk boxes to be used in production. I know it's being done because as far as I know Twilio is using Amazon for their Asterisk boxes. I have read all the fun articles on building hobby type systems and that's all great. What I really need to hear is from those that have deployed Asterisk in Amazon or Digital Ocean and how many simultaneous calls they are pushing through it and what the call quality and reliability has been. Right now I am still using dedicated hardware but I could become much more redundant and scale much faster using Amazon or Digital Ocean. Thanks in advance for any information from those that have already been down this road... -- _ -- 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 Alternatives
[asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
Took me a while but I have finally embraced cloud computing and all the benefits. The only thing I have yet to feel comfortable about putting in the cloud is real live Asterisk boxes to be used in production. I know it's being done because as far as I know Twilio is using Amazon for their Asterisk boxes. I have read all the fun articles on building hobby type systems and that's all great. What I really need to hear is from those that have deployed Asterisk in Amazon or Digital Ocean and how many simultaneous calls they are pushing through it and what the call quality and reliability has been. Right now I am still using dedicated hardware but I could become much more redundant and scale much faster using Amazon or Digital Ocean. Thanks in advance for any information from those that have already been down this road... -- _ -- 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] Make phone ring through webserver using Asterisk
What do you want to happen once the call is made? You can choose to fire the call off using the originate command with the Asterisk Manager Interface from a PHP page or some other similar language. No need for Perl on the Asterisk box at all really unless you need it for something else. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate Date: Sat, 16 Nov 2013 16:53:59 +0530 From: omakhileshch...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Make phone ring through webserver using Asterisk What is the easiest way? And how can it be implemented? I thought to something like: I request a page to the webserverPerl sends to asterisk a number to dial (Perl and asterisk are running in the same machine) Asterisk calls the phoneor A Perl sip client registers to remote asterisk serverPerl sip client sends to asterisk the number to dialPhone ringsi don't care if i can hear something, it's enough that it rings -- _ -- 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] chan_sip.c:9602 copy_header: No field 'CSeq' present to copy
Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this. -- _ -- 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] Pull call out of queue
Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call parking but, I think parking is more than I need and it potentially introduces more complications. I will be doing this through the manager interface on Asterisk 1.8.x. Any ideas, thoughts or help would be greatly appreciated. Thanks in advance for any help. -- _ -- 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] Local agent/member in-use after transfer
I currently have all agents/members logged in with local channels. When a call is sent to one of the agents, then the agent transfers the call out the line frees up on their phone but still shows in-use until the call that was transferred is hung up. How can I free up the agent/local channel when the call is transferred? This is a huge problem because the agent can no longer receive calls on their extension. If they are the only agent logged in, then no other calls can be answered. If the transferred calls last an hour then no calls can be answered by this agent for an hour. I know I can set ringinuse=yes but this causes the agent to be interrupted while on calls which is not the desired result. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local agent/member in-use after transfer
From: tjrl...@live.com To: asterisk-users@lists.digium.com Date: Thu, 1 Aug 2013 12:50:32 -0500 Subject: [asterisk-users] Local agent/member in-use after transfer I currently have all agents/members logged in with local channels. When a call is sent to one of the agents, then the agent transfers the call out the line frees up on their phone but still shows in-use until the call that was transferred is hung up. How can I free up the agent/local channel when the call is transferred? This is a huge problem because the agent can no longer receive calls on their extension. If they are the only agent logged in, then no other calls can be answered. If the transferred calls last an hour then no calls can be answered by this agent for an hour. I know I can set ringinuse=yes but this causes the agent to be interrupted while on calls which is not the desired result. -- _ -- 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 I found a solution and wanted to post it for those that may run into this trouble in the future. I use the manager interface to login my agents using a web page. After much digging I finally found the StateInterface: option available in 1.6 and above. I added it to my PHP login screen like this.. fputs($socket2, StateInterface: SIP/.$agentid.\r\n); The problem is that the queue was monitoring the local channel in terms of when a call was hungup or not, allowing other calls to come through. When a transfer happened the Local channel was not released. Adding the StateInterface option apparently allows the queue to monitor the actual channel, not the local channel. I couldn't find much documentation on this option, just stumbled upon it. Fixed my issue though! Thought I would add to the little info that seems to be out there about this option. -- _ -- 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] Ignoring MOH directory and using default
It was the darn restart, thanks! I wasn't aware that musiconhold.conf was only read at start but now I remember that Asterisk only reads certain files upon reload. Thx On Jul 31, 2009, at 2:03 PM, Danny Nicholas da...@debsinc.com wrote: Are your permissions ok (files and directories)? Did you restart asterisk after modifying moh.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Friday, July 31, 2009 1:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Ignoring MOH directory and using default Wow, been a long time since I have been on the list.. A few years to be exact :) Glad to be back in the land of Asterisk.. I have a box running Asterisk 1.4.8 that's been real solid and I have a bunch of custom stuff running on it. I am trying to move this to a new piece of hardware and everything is going well but I am having MOH issues. Basically, I use raw sound files for my music on hold on the old machine and I want to do the same on the new machine. I installed the same version of asterisk on the new box, then moved all my config files and sound files over. For some odd reason, the music on hold that Asterisk is playing is the default stuff in the default moh directory even though I have the class below defined in musiconhold.conf. The directory and sound files do exist and all this works fine on the old box, same config files, same sound files, same path, same Asterisk version. Any ideas? Thanks! #musiconhold.conf## [default] mode=files ;directory=/var/lib/asterisk/moh directory=/var/lib/asterisk/mohtjr ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Conference Bridge?
Can Asterisk act as a SIP conference bridge? Looking through the source I notice that it's required to have a Zaptel interface installed. Why is this a requirement? Can you not mesh the VoIP streams together? Thanks, T..S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
