Re: [asterisk-users] Delay when dialing...
Hi there, I can confirm that this is indeed the problem. If you follow the advise below you will be sorted. From my mobile phone On 23 Jul 2021, 8:44 am, at 8:44 am, Jean Aunis wrote: >Le 22/07/2021 à 18:32, Carlos Chavez a écrit : >> I started noticing a few days ago that whenever I dial any number > >> or extension there is a delay of 5 to 10 seconds before Asterisk >> reacts. I see nothing on the CLI for that time and then the call >goes >> through. I have checked my network to make sure there is nothing >> slowing down packets between the phones and the server. >> >> Any settings I should check on the Asterisk side? This is >> happening with all phones (several brands). >> >Hi, > >I've seen this problem several times when there is no DNS resolution of > >Asterisk's hostname. > >Try to add your hostname to /etc/hosts and check if it's better. > >Regards, > >Jean > > > >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >Check out the new Asterisk community forum at: >https://community.asterisk.org/ > >New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pass Sound files as Argument to Macro Asterisk 1.8
Hi All, I was wondering if it is possible to pass sound files to a macro as an argument in Asterisk 1.8? Thanks! Regards, Daniel van den Berg SureTel South Africa 087-944-7873 -- _ -- 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] Want Queues to ignore mobile operators voice mails and continue ringing...?
Hi All, Lets say I want to setup a queue that will handle inbound calls to dynamically added agents that are all mobile numbers. Now when I do this setup it works, it loads the agents dynamically and if the mobile phone is on and have reception it works. But when the phone is for arguments sake off or dont have reception it goes to voice mail for that mobile phone. I don't want this to happen...:) I would like for the queue to continue ringing until there is a time out specified which then takes the caller out of the queue and to voice mail which I then intend to mail somewhere. I guess my question is can this be done in Asterisk? Can I force clients in this queue not to leave a voice message on the mobile phone but rather the Asterisk system? Because when the mobile phone which is an agent in the queue goes to voice mail it answers the call and then plays the voice mail message. My initial thoughts are to maybe ask the mobile operator to switch off the voice mail functionality on those mobile phones and rather give a busy or engaged tone, but I would rather want to do this in Asterisk. Any help or advise on this matter will be greatly appreciated. Thanks! Daniel van den Berg SureTel - South Africa -- _ -- 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] OT - Merry Christmas and a Happy and Prosperous 2014
Hi Josue, May God the Father of our Lord and Saviour Jesus Christ bless you and your loved ones always! On 12/24/2013 10:37 PM, Josué Conti wrote: > Dear Ladies and Gentlemen how are you? > I would like to wish everyone and all their families, may God continue > to bless and always illuminating your steps, that this Christmas the > joy of our Lord Jesus invade your home and participate in their family > on this Christmas. > May God bless you always! > Merry Christmas and a Happy and Prosperous 2014. > > > With Best Regards! > > Josue > > -- _ -- 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] Tired of dropouts and garbled phone calls - where to go next?
Hi there, In other words you are maybe on 60ms and they are on 20ms or vice versa. Do a wireshark trace and see if the codecs and ptime agree on both sides otherwise you will get grabbled sounds. On 10/29/2013 02:49 PM, Daniel van den Berg wrote: > Hi there, > > Sounds like codec ptime mismatch...what codec are you using? If you > are using g729 make sure that you and your provider is giving the same > ptime. > > On 10/29/2013 11:55 AM, Stelios Koroneos wrote: >> On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: >>> All, >>> >>> >>> The users in our organization are well, quite frankly, sick of phone >>> service that is being provided. The choppy phone calls, and drop outs >>> are detrimental to our sales force. >>> >>> >>> I've tried about everything I can think of. >>> >>> >>> Moved the asterisk server from VM machine to dedicated machine >>> More than enough bandwidth >>> Setting 802.1p = 7 >>> Set Dedicated voice traffic 35% of bandwidth. >>> >>> >>> Not sure what option would be the best >>> >>> >>> Put analog lines in the conference room to avoid the dropouts >>> - leave the sip lines in place for day to day use >>> Hire a consultant >>> Ditch the system and buy a pre-packaged system - RingCentral >>> or some such. >>> >>> >>> There are no local asterisk professionals who can help, and we are a >>> little leery of opening up our system to outside consultants. >>> >>> >>> Anyone else face the above, and finally abandoned Asterisk for a >>> commercial system? >>> >>> >>> We have 167 users. >>> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the >>> conference rooms. >>> >>> >>> Suggestions welcome. >>> >>> >> A general rule of thump after several years with voip >> >> Voip turns out to be the "canary in the coal-mine" of a network. The >> smallest change or problem will manifest itself as a voip issue no >> matter what. >> >> >> Now to some practical advice >> >> Voip was designed for LAN's, The moment voip packets leave your lan and >> go into a WAN of any sort, it could be the source of frustration for >> many reasons. >> >> 1) Lots of routers/modems are not build to handle intense voip traffic. >> voip generates lots of small in size UPD packages. In most of the cases >> the routers/modems bridging your lan with the wan have no problem >> handling them BUT what i have found is that once you get over a >> threshold of traffic its possible the routers/modem can not cope with >> it, mainly because the large number of packets they have to process. >> In most enterprise grade routers the specs give you 2 numbers for the >> size of data the router can handle. >> total throughput and pps (packets per second). >> Usually total throughput is calculated using a packet size of around >> 1500bytes and it takes the router the same resources to process a 1500 >> bytes package as it does a 90bytes packet of a g729 call, as it just >> looks at the headers and not the payload.So yes your router can handle >> 60Mbits (of 1500byte frames) which is about 5000 packers per second but >> for voip that translates to less than 4Mbits of data (5000 packets of 90 >> bytes) >> I think you can get the picture >> >> >> 2) Because of 1) its possible that your ISP has issues, especially if >> its handling lots of voip traffic while its equipment is not optimized >> for that. >> >> >> 3) QOS and queing in general >> Whatever you do with QOS to get a better priority/quality, the dirty >> secret is, you can only control what YOU send, not what you receive. >> And even that is true till your modem/router. Once the packet is gone >> you have no control of how it will be handle by all intermediates till >> it reaches its destination. >> You have no idea if qos is honored by ALL hops and what kind of queuing >> they apply (if they do) to that port/service/qos mark >> That beeing said, its possible that you *might* have much better luck >> with sip and sip rtp than with iax rtp if your isp and all its >> interconnects bother to offer qos for rtp. >> Now for receiving it can be even harder if your isp does not provide >> correct priority queuing for the rtp
Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
Hi there, Sounds like codec ptime mismatch...what codec are you using? If you are using g729 make sure that you and your provider is giving the same ptime. On 10/29/2013 11:55 AM, Stelios Koroneos wrote: > On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: >> All, >> >> >> The users in our organization are well, quite frankly, sick of phone >> service that is being provided. The choppy phone calls, and drop outs >> are detrimental to our sales force. >> >> >> I've tried about everything I can think of. >> >> >> Moved the asterisk server from VM machine to dedicated machine >> More than enough bandwidth >> Setting 802.1p = 7 >> Set Dedicated voice traffic 35% of bandwidth. >> >> >> Not sure what option would be the best >> >> >> Put analog lines in the conference room to avoid the dropouts >> - leave the sip lines in place for day to day use >> Hire a consultant >> Ditch the system and buy a pre-packaged system - RingCentral >> or some such. >> >> >> There are no local asterisk professionals who can help, and we are a >> little leery of opening up our system to outside consultants. >> >> >> Anyone else face the above, and finally abandoned Asterisk for a >> commercial system? >> >> >> We have 167 users. >> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the >> conference rooms. >> >> >> Suggestions welcome. >> >> > A general rule of thump after several years with voip > > Voip turns out to be the "canary in the coal-mine" of a network. The > smallest change or problem will manifest itself as a voip issue no > matter what. > > > Now to some practical advice > > Voip was designed for LAN's, The moment voip packets leave your lan and > go into a WAN of any sort, it could be the source of frustration for > many reasons. > > 1) Lots of routers/modems are not build to handle intense voip traffic. > voip generates lots of small in size UPD packages. In most of the cases > the routers/modems bridging your lan with the wan have no problem > handling them BUT what i have found is that once you get over a > threshold of traffic its possible the routers/modem can not cope with > it, mainly because the large number of packets they have to process. > In most enterprise grade routers the specs give you 2 numbers for the > size of data the router can handle. > total throughput and pps (packets per second). > Usually total throughput is calculated using a packet size of around > 1500bytes and it takes the router the same resources to process a 1500 > bytes package as it does a 90bytes packet of a g729 call, as it just > looks at the headers and not the payload.So yes your router can handle > 60Mbits (of 1500byte frames) which is about 5000 packers per second but > for voip that translates to less than 4Mbits of data (5000 packets of 90 > bytes) > I think you can get the picture > > > 2) Because of 1) its possible that your ISP has issues, especially if > its handling lots of voip traffic while its equipment is not optimized > for that. > > > 3) QOS and queing in general > Whatever you do with QOS to get a better priority/quality, the dirty > secret is, you can only control what YOU send, not what you receive. > And even that is true till your modem/router. Once the packet is gone > you have no control of how it will be handle by all intermediates till > it reaches its destination. > You have no idea if qos is honored by ALL hops and what kind of queuing > they apply (if they do) to that port/service/qos mark > That beeing said, its possible that you *might* have much better luck > with sip and sip rtp than with iax rtp if your isp and all its > interconnects bother to offer qos for rtp. > Now for receiving it can be even harder if your isp does not provide > correct priority queuing for the rtp stream, as latencies can build fast > especially on "busy hours" (which happen to be the same hours people use > their phones the most...) where people download stuff,emails etc. > > ping.icmp and all the other networking monitoring tools/protocols could > be an indicator BUT its most probable that they will be handled by the > isp and its interconnects at the higher qos priority > The only way to see how rtp traffic is handled is to run rtp traffic. > > The only way around this is a "dedicated circut" MPLS or similar between > the points of interest (i.e offices), with specific SLA which usually > means much much higher costs. >> > Finally my 2 cents for troubleshouting. > Check the network first ! > Find what triggers the problem. > Is it something that happens all time regardless of traffic ? > is it periodic ? (when bw goes over X percent, or at a specific time of > day ?) > Try different qos settings/priority queuing on the router > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digita
Re: [asterisk-users] How to bind to ipv4 & ipv6
Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 & IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 & IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 05:59 PM, Johan Wilfer wrote: > http://lists.digium.com/pipermail/asterisk-users/2013-March/278122.html > > Google :-) > > /J > > 2013-09-27 17:47, Daniel van den Berg skrev: >> Hi Asghar, >> >> How do I search the site as I dont see a search bar anywhere...could you >> please give me the link to the solution in the list or educate me on how >> to search the site bar going through every thread one by one. :) >> >> Thanks! >> >> Regards, >> On 09/27/2013 04:43 PM, Asghar Mohammad wrote: >>> Hi, >>> Please Search the List there is already a post and solution. >>> >>> >>> >>> On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg >>> mailto:aster...@suretel.co.za>> wrote: >>> >>> Hi All, >>> >>> This is my 1st post so lets go. >>> >>> What I need to achieve is the following. I have server with both >>> IPv4 >>> addresses and IPv6 addresses. The problem that I am encountering >>> is that >>> in the sip.conf I am having difficulties to bind to both the >>> IPv4 and >>> IPv6 addresses. >>> >>> Can someone please assist me in this regard as I need to connect >>> another >>> server to this server on IPv6 while the rest of the clients are >>> connecting on IPv4. >>> >>> I need to know how to get this working? >>> >>> -- >>> >>> _ >>> -- 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] user list archive
Sorry :) On 09/27/2013 06:16 PM, Rusty Newton wrote: > On Fri, Sep 27, 2013 at 11:14 AM, Daniel van den Berg > wrote: >> Hi All, >> >> I dont really see a solution there to the problem, just that the matter >> was discussed? >> >> Can Asterisk or can it not listen for IPv4 & IPv6 addresses at the same >> time? I only see that there is mention that you must use the bindaddr=:: >> for it to listen for IPv4 & IPv6 but when I do this my IPv4 connections >> drops. >> > You responded to the wrong thread. oops! :) > -- _ -- 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] user list archive
Hi All, I dont really see a solution there to the problem, just that the matter was discussed? Can Asterisk or can it not listen for IPv4 & IPv6 addresses at the same time? I only see that there is mention that you must use the bindaddr=:: for it to listen for IPv4 & IPv6 but when I do this my IPv4 connections drops. Thanks! On 09/27/2013 06:10 PM, Rusty Newton wrote: > On Fri, Sep 27, 2013 at 10:20 AM, Paul Albrecht wrote: >> What's up with the user list archive? It hasn't been updated since the 23rd. > We did some Mailman maintenance on the 23rd, some configuration may > have been goofed up. We'll look into it. 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] How to bind to ipv4 & ipv6
Hi Asghar, How do I search the site as I dont see a search bar anywhere...could you please give me the link to the solution in the list or educate me on how to search the site bar going through every thread one by one. :) Thanks! Regards, On 09/27/2013 04:43 PM, Asghar Mohammad wrote: > Hi, > Please Search the List there is already a post and solution. > > > > On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg > mailto:aster...@suretel.co.za>> wrote: > > Hi All, > > This is my 1st post so lets go. > > What I need to achieve is the following. I have server with both IPv4 > addresses and IPv6 addresses. The problem that I am encountering > is that > in the sip.conf I am having difficulties to bind to both the IPv4 and > IPv6 addresses. > > Can someone please assist me in this regard as I need to connect > another > server to this server on IPv6 while the rest of the clients are > connecting on IPv4. > > I need to know how to get this working? > > -- > _ > -- 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] How to bind to ipv4 & ipv6
Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses. The problem that I am encountering is that in the sip.conf I am having difficulties to bind to both the IPv4 and IPv6 addresses. Can someone please assist me in this regard as I need to connect another server to this server on IPv6 while the rest of the clients are connecting on IPv4. I need to know how to get this working? -- _ -- 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
