[asterisk-users] Running safe_asterisk
About two weeks ago there was a thread about asterisk suddenly dying - I posted a response that the same happens to my asterisk about once a month, sometimes more. Someone suggested using 'safe_asterisk' (and get hold of a core dump) which sounds like a good idea, but one thing I can't figure is how to get "module reload app_queue" executed automatically at startup? /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On 22/02/10 16:18, --[ UxBoD ]-- wrote: > Hi, > > looking for your valued input on suitable suggestions for high quality VoIP > DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and > looking to a new manufacturer. > Another vote for the Siemens Gigaset range. Been using the S685IP almost since the day it was released here in the UK. Nice handsets, great voice quality, but as others have said the UI can be a bit slow. Do watch out for the last firmware release though. Siemens had some trouble with that although I personally haven't experienced the issues reported: http://www.mgraves.org/voip/2009/11/gigaset-firmware-update-released/ It looks like a new firmware release is imminent: http://www.mgraves.org/voip/2010/02/gigaset-news-new-beta-firmware-release/ HTH Alan -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News) wrote: > Another vote for the Siemens Gigaset range. Been using the S685IP almost > since the day it was released here in the UK. Nice handsets, great voice > quality, but as others have said the UI can be a bit slow. Alan, don't forget the link to the discussion on your excellent site: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Same experience here, we've been running our 2-person soho for a couple of years with one base and 2 S675IP handsets. - They look and act like a "regular" cordless phone to the average person who is not a telephony geek. - They work well with a bunch of SIP accounts and g729 if you have that possibility - common headset jack works with cheap headsets - landline connection that works transparently when the Internet connection is down - simple dialplan to route calls - Excellent battery life and talk time Ours have performed flawlessly. Yes, the interface is slow and so is the phone menu system. We just purchased another base and handset for our new office. I love this phone and wish I was getting a commission on the number of units I've probably "sold". /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On Mon, 22 Feb 2010, Gordon Henderson wrote: > On Mon, 22 Feb 2010, --[ UxBoD ]-- wrote: > >> Hi, >> >> looking for your valued input on suitable suggestions for high quality >> VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk >> 1.6 and looking to a new manufacturer. > > Siemens Gigaset over M3's anyday. Nicer displays, bigger handsets and > buttons. > > The downside is that they are slow - both on the handsets and their web > interface - make sure you're using a browser with fast javascript - e.g. > Chrome. And following up my own reply - I've found the newer ones to be better in terms on browser speed than the original 450IP's were. And I'm going to use some in a local charity/museum shortly too (A580's) - as audio guides for people who want to self-guide themselves - since you can have 2 independant (VoIP) audio streams on a base-station, it's almost economical - I'll be providing 2 bases, 4 handsets and a small embedded asterisk box to hold the audio files - punters will just dial the extension of the room they're in to get the commentary (from a crib-sheet, or label on an artifact) ... The commercial ones I've seen seem to store the audio inside the handset, but DECT will cover our area nicely - the commercial ones are also large, presumably to stop people wandering off with them... Hope that won't be a problem for us! Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup
Hello, worst aspect is that - if SIP clients do not have such a timeout, and in that case if killing an asterisk and to start it up again - so it is nothing to do with this asterisk timeout. Regards, On 23 February 2010 08:44, Olle E. Johansson wrote: > > 23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson: > > > On 100222 1313, JT wrote: > >> When a SIP device dials another SIP device...Asterisk connects the calls > and > >> displays the channel information. > >> If one of those SIP devices hangs up, Asterisk receives the hangup > notice > >> and disconnects the call/channel. > >> However - what does Asterisk do when the network cable is unplugged from > one > >> of the SIP devices...?! > > > > Jared already mentioned SIP session timers, which are supported starting > with 1.6. Here's my experience. While I am running 1.6, the software stack > that is used for agent softphone (PJSIP) does not support the session > timers. If the softphone crashes in a call, the call would get stuck exactly > as you describe. > > > > I am working around this problem by setting rtp timeouts in sip.conf: > > > > [general] > > rtptimeout=10 > > rtpholdtimeout=300 > > > > This means that if RTP flow stops while the agent is in the call, the > call will be disconnected in 10 seconds. If the call was put on hold by the > agent, it will be disconnected in 300 seconds. Your timeouts may vary. > > > > The caveat here is that it is perfectly normal NOT to transmit any RTP > data in case of long silence. > Not in Asterisk - we do not really support silence suppression. The > recommendation is to turn it off on the phones. > > > This is why the SIP timers were introduced in the first place: there is > no correct way to detect when the client is going away, as no activity is a > good session state. > > > > I am able to get away with the small timeout because I set the PJSIP > client to always transmit RTP, by turning off voice activity detection > feature (VAD). If you want to support that feature, set rtptimeout as high > as for how long you allow absolute silence on the line without disconnecting > it. > > Just to complete this discussion - we also have the absolute timeout that > is a lifesaver in many cases. If you set this to a time that's larger than > the normal calls, Asterisk will hang up the call. I very often set it to two > hours, just to make sure that if anything strange happens, all calls will be > cancelled out at some point. > > /O > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- We never did too much talking anyway So don't think twice, it's all right -- There are more things in heaven and earth, Horatio, Than are dreamt of in your philosophy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On 23/02/10 08:38, Randy R wrote: > On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News) > wrote: >> Another vote for the Siemens Gigaset range. Been using the S685IP almost >> since the day it was released here in the UK. Nice handsets, great voice >> quality, but as others have said the UI can be a bit slow. > > Alan, don't forget the link to the discussion on your excellent site: > > http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Thanks for the plug Randy :-) Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
- "Philipp von Klitzing" wrote: > Hi! > > > looking for your valued input on suitable suggestions for high > quality > > VoIP DECT phones. I am having real issues with my Snom M3s and > Asterisk > > 1.6 and looking to a new manufacturer. > > Define high quality. > Anyone here used any of these below with Asterisk? > > * NEC AP300 and NEC DECT C124 or NEC DECT M155 > * Aastra RFP L32 with Aastra 142 DECT or Aastra 610d/620d DECT > > I am really curious about those, especially the M155. > Philipp High quality to me means well built, reliable, good protocol support and above all a responsive manufacturer. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redirect question
Hello, I am relative new to Asterisk and we want the following: ExternalCall-->UserPBX-->DialOutNormal | ^ V | Asterisk | ^ V | Application We have the above configuration and we would like to tell the UserPBX to set the call from the ExternalCall directly through to DialOutNormal in such a manner, that Asterisk and Application are free again. If the UserPBX cannot handle such a request the Asterisk will do the DialOut. I know we need a trusted relation between the UserPBX and the Asterisk, but what command do we need to instruct the UserPBX to set the original call from ExternalCall through to DialOutNormal? Kind regards, Bert Mengeribk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balance outgoing calls
Thank you Steve, that's a good idea.
If I use a global variable like
--> IF GLB > 2 GLB = 0
dial(iax2/isp${GLB}/${EXTEN})
--> GLB = GLB +1
I believe this could cause a race condition if two calls are sent to
the carrier at the same time?
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Re: [asterisk-users] Redirect question
On 23 Feb 2010, at 09:58, Bert Mengerink wrote: > I know we need a trusted relation between the UserPBX and the > Asterisk, but what command do we need to instruct the UserPBX to set > the original call from ExternalCall through to DialOutNormal? Ask whoever made 'UserPBX'? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redirect question
Hi Steve, UserPBX could be any brand PBX, like Ericson, Avaya, etc. Or even another asterisk. Therefor I added the provision, that the UserPBX should support such a strategy. Is there a general SIP command to provide the action we want? Kind regards, Bert -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: dinsdag 23 februari 2010 11:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Redirect question On 23 Feb 2010, at 09:58, Bert Mengerink wrote: > I know we need a trusted relation between the UserPBX and the > Asterisk, but what command do we need to instruct the UserPBX to set > the original call from ExternalCall through to DialOutNormal? Ask whoever made 'UserPBX'? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.435 / Virus Database: 271.1.1/2702 - Release Date: 02/22/10 19:34:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX devices not registering after upgrade to asterisk 1.4.29
Hi All, We have encountering issue that IAX enable voice gateways not registering with asterisk after upgrade from asterisk 1.4.18.1 -> 1.4.29 Before that IAX works very well. If any one have similar issue and solution for that let me know. -- Thanks & Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running safe_asterisk
On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote: > About two weeks ago there was a thread about asterisk suddenly dying - I > posted a response that the same happens to my asterisk about once a > month, sometimes more. > Someone suggested using 'safe_asterisk' (and get hold of a core dump) > which sounds like a good idea, but one thing I can't figure is how to > get "module reload app_queue" executed automatically at startup? All modules are loaded at startup. Why would you need a reload? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple instances of Asterisk on the same host...
On Mon, Feb 22, 2010 at 11:23:29PM +, Gordon Henderson wrote: > On Mon, 22 Feb 2010, Roderick A. Anderson wrote: > > > Gordon Henderson wrote: > >> Interesting thread recently about virtual servers... > >> > >> I'm thinking of doing something similar - right now looking at Containers > >> (lxc) rather than "proper" virtualisation though, however it got me > >> thinking of a "poor mans virtualisation" solution... > >> > >> This would assume you have a real server to start with and full root > >> access... > >> > >> I was thinking of simply running multiple asterisks on the same box, each > >> with their own /etc/asterisk config directory (in e.g. > >> /home/v1/etc/asterisk, /home/v2/etc/asterisk and so on - obviously give > >> them unique /home/v1/spool/asterisk/ , etc. directories too, but for the > >> most part things like /var/lib/asterisk/sounds and modules can be shared. > >> (exception being astdb!) It just means a custom > >> /etc/asterisk/asterisk.conf file for each instance and asterisk being > >> started with the correct config file - /home/v1/etc/asterisk.conf, etc. > >> > >> So giving each asterisk it's own IP address (eth0:1, eth0:2, etc.) and > >> changing the bindaddr parameter in each one to suit multiple IP addresses > >> bound to the 'host' would seem to be the way to do it - each asterisk can > >> still use ztdummy/dhadidummy for timing if required (or does it stop > >> multiple asterisks opening it?) > >> > >> Anyone done this or contemplated doing it? > > > > I have heard of a company, name completely escapes me right now, that > > appears to use Linux-Vserver. > > > > I am trying to find the time to move my business system to a > > Linux-Vserver from a Micro-Linux Asterisk Server and the only issue I'm > > aware of is DAHDI/ZAPTEL might have to be run in the "host" instead of > > the guests. Then some permissions set so the guests can access it DAHDI. > > My aim is to actually use LXC as it has kernel level support (as of > 2.6.29) and will be supported by most distros soon if not already. > Linux-Vserver appears to be depreciated by at least Debian, probably > Ubuntu too, but I've no idea about the world of Red Hat/Fedora/Centos, > etc.. I tried OpenVZ, but it seems to have even poorer support, and no > updated for some time either. Actually: Linux-VServer is deprecated much in favour of OpenVZ. The OpenVZ developers have been much more willing to work with the upstream kernel maintainers. But then again, lxc uses much of the work on containers done also by and for OpenVZ. Sort of like the VMWare/Xen/KVM story all over again, with lxc playing the role of KVM. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
On Tue, Feb 23, 2010 at 10:50 AM, --[ UxBoD ]-- wrote: > High quality to me means well built, reliable, good protocol support and > above all a responsive manufacturer. Incidentally, I've dropped two of the S675IP handsets on the hardwood floor a few times, still working fine. Concrete may be a different matter. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running safe_asterisk
Tzafrir Cohen wrote: > On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote: >> About two weeks ago there was a thread about asterisk suddenly dying >> - I posted a response that the same happens to my asterisk about once >> a month, sometimes more. >> Someone suggested using 'safe_asterisk' (and get hold of a core dump) >> which sounds like a good idea, but one thing I can't figure is how to >> get "module reload app_queue" executed automatically at startup? > > All modules are loaded at startup. Why would you need a reload? > To be honest I don't remember any more, I just know my queueing doesn't work unless I reload. I think it's a timing issue at startup - that app_queue gets loaded too early or something. ah, here is my question about the same, but back in 2007: http://lists.digium.com/pipermail/asterisk-users/2007-May/188072.html /Per Jessen, Zürich -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec translation in Asterisk
Hi Group, Can anybody explain me in detail how the codec translation happens on asterisk side when 2 endpoints have different codecs? Thanking you in advance. --SM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls per second limit in manager
My dear friend Matt Riddell insists that the Manager only can dial 5 calls per seconds, which I find ridiculous. Is there a way to prove him wrong and have him lift the limit that has been plaguing the life of us users of SineDialer and SmoothTorrque Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Denying call transfer to certain extensions
But this won't help if 100 or 101 wants to call 102. What I want is, if a call coming from a trunk 100 rings, and if the caller wants to be transfered to 101, the transfer is denied. In other words, 101 can't get transfered calls. Danny Nicholas wrote: > Follow-me will most likely be your best bet for this trick. Say you have > extensions 100, 101 and 102. 100 is the receptionist, 101 is sales and 102 > is the boss, who doesn't want to be disturbed. If you set up followme on > 102 to go to voicemail or whatever, 102 won't ring. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Ossama > Sent: Monday, February 22, 2010 5:53 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Denying call transfer to certain extensions > > Hi all, > > Is there a way to deny call transfers to certain extensions? > > Thanks, > Ahmed Ossama > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple instances of Asterisk on the same host...
On Tue, 23 Feb 2010, Tzafrir Cohen wrote: >> My aim is to actually use LXC as it has kernel level support (as of >> 2.6.29) and will be supported by most distros soon if not already. >> Linux-Vserver appears to be depreciated by at least Debian, probably >> Ubuntu too, but I've no idea about the world of Red Hat/Fedora/Centos, >> etc.. I tried OpenVZ, but it seems to have even poorer support, and no >> updated for some time either. > > Actually: Linux-VServer is deprecated much in favour of OpenVZ. The > OpenVZ developers have been much more willing to work with the upstream > kernel maintainers. Is it? I was very frustrated that OpenVZ's latest 'stable' release was for 2.6.18 and trying to patch it into a current keren was nigh-on impossible. I even tried the Debin kernels with their patches but in all cases it produced a kernel that would not boot, so I gave up. LXC is supported in the kernel without any patches and it now working well for me, so I'm sticking to it. > But then again, lxc uses much of the work on containers done also by and > for OpenVZ. Sort of like the VMWare/Xen/KVM story all over again, with > lxc playing the role of KVM. And LXC got into the kernel before the others - what that means is anyones guess - probably because it was sponsored/written by IBM? Actually, I'm quite impressed by it so-far - I've had no need to look at virtual stuff for a while, but did some investigations recently for another project and the whole idea of Containers is growing on me rapidly... I get a feeling that it's going to be better suited for running things like asterisk than full-on virtualisation might be. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Asterisk 1.6 and DECT Phones
- "Philipp von Klitzing" wrote: > Hi! > > > looking for your valued input on suitable suggestions for high > quality > > VoIP DECT phones. I am having real issues with my Snom M3s and > Asterisk > > 1.6 and looking to a new manufacturer. > > Define high quality. > Anyone here used any of these below with Asterisk? > > * NEC AP300 and NEC DECT C124 or NEC DECT M155 > * Aastra RFP L32 with Aastra 142 DECT or Aastra 610d/620d DECT > > I am really curious about those, especially the M155. > Philipp Hmmm, they do look very interesting indeed! -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple instances of Asterisk on the same host...
On Tue, Feb 23, 2010 at 12:19:31PM +, Gordon Henderson wrote: > On Tue, 23 Feb 2010, Tzafrir Cohen wrote: > > But then again, lxc uses much of the work on containers done also by and > > for OpenVZ. Sort of like the VMWare/Xen/KVM story all over again, with > > lxc playing the role of KVM. > > And LXC got into the kernel before the others - what that means is anyones > guess - probably because it was sponsored/written by IBM? KVM was not sponsored by any big-name company (RedHat, IBM and the others had their bets on Xen, at the time). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S card
On 22 February 2010 14:07, Razza wrote: > I'm using CentOS5.4, can anyone advise how I can make DAHDi work with > a generic HFC-S card? On 22 February 2010 15:12, Pedro Santos wrote: > I´m using centos 4.8 server, and i don't now how integrate zaphfc with dadhi Anyone able to help, with some simple advice? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S card
On Tue, Feb 23, 2010 at 12:48:15PM +, Razza wrote: > On 22 February 2010 14:07, Razza wrote: > > I'm using CentOS5.4, can anyone advise how I can make DAHDi work with > > a generic HFC-S card? > > On 22 February 2010 15:12, Pedro Santos wrote: > > I´m using centos 4.8 server, and i don't now how integrate zaphfc with dadhi > > Anyone able to help, with some simple advice? Have you managed to install those zaphfc drivers? Those are basically the same ones from http://code.google.com/p/zaphfc/ , and the same ones in packages from Elastix and Debian. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S card
On 23 February 2010 12:58, Tzafrir Cohen wrote: > Have you managed to install those zaphfc drivers? > > Those are basically the same ones from http://code.google.com/p/zaphfc/ Hi Tzafrir. I checkout out that but there were no instructions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] directrtp with SIP + H.323
We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX devices not registering after upgrade to asterisk 1.4.29
Hi! > We have encountering issue that IAX enable voice gateways not > registering with asterisk after upgrade from asterisk 1.4.18.1 -> 1.4.29 > Before that IAX works very well. If any one havesimilarissue and solution > for that let me know. Search or google for "calltokenoptional". http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf http://downloads.asterisk.org/pub/security/AST-2009-006.html Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] directrtp with SIP + H.323
-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Michelle Dupuis skrev: > We're creating a SIP gateway for a client that will take one leg of a > call in via SIP, and out the other side via H.323. To minimize load on > the gateway, we would like to have the RTP stream bypass the gatewayy > altogether (directrtp/reinvite). Is this possible with these to protocols? Unfortunately, that is not possible. - - Tommy -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuD42MACgkQ573V05EH/pbtrQCfY4ojpCKo6oTmKerJiB+s/14l qMAAn2PAmz2qCJI+W0EPqk8Khn9K1UKx =hrY5 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] directrtp with SIP + H.323
Tommy Botten Jensen wrote: > Michelle Dupuis skrev: >> We're creating a SIP gateway for a client that will take one leg of a >> call in via SIP, and out the other side via H.323. To minimize load on >> the gateway, we would like to have the RTP stream bypass the gatewayy >> altogether (directrtp/reinvite). Is this possible with these to protocols? > > Unfortunately, that is not possible. As I understand it, the H.323 protocol, in most implementations, does not allow redirecting the media endpoints after the call is setup. In a pure proxy-type environment, where the media never goes through a switch at all, this would be possible, but for a B2BUA like Asterisk, it's not likely to be possible. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] directrtp with SIP + H.323
On Tue, 2010-02-23 at 08:22 -0500, Michelle Dupuis wrote: > We're creating a SIP gateway for a client that will take one leg of a > call in via SIP, and out the other side via H.323. To minimize load > on the gateway, we would like to have the RTP stream bypass the > gatewayy altogether (directrtp/reinvite). Is this possible with these > to protocols? > > Thanks > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users IMHO, It's impossible ;) -- Best regards, Vince Mallow xmpp: w...@jabber.slan.ru web: http://gentoo-way.blogspot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running safe_asterisk
On Tuesday 23 February 2010 05:27:55 Per Jessen wrote: > Tzafrir Cohen wrote: > > On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote: > >> About two weeks ago there was a thread about asterisk suddenly dying > >> - I posted a response that the same happens to my asterisk about once > >> a month, sometimes more. > >> Someone suggested using 'safe_asterisk' (and get hold of a core dump) > >> which sounds like a good idea, but one thing I can't figure is how to > >> get "module reload app_queue" executed automatically at startup? > > > > All modules are loaded at startup. Why would you need a reload? > > To be honest I don't remember any more, I just know my queueing doesn't > work unless I reload. I think it's a timing issue at startup - that > app_queue gets loaded too early or something. ah, here is my question > about the same, but back in 2007: > > http://lists.digium.com/pipermail/asterisk-users/2007-May/188072.html You need to load the chan_local.so channel before pbx_config.so loads, so that your Local channels have the right devicestate. Adding 'preload => chan_local.so', followed by 'preload => pbx_config.so', to your /etc/asterisk/modules.conf should be sufficient. Otherwise, when app_queue queries the state of the Local channels at startup, it finds them to be unavailable. This is why reloading app_queue.so fixes the issue. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balance outgoing calls
On Tue, 23 Feb 2010, Alejandro Recarey wrote:
> If I use a global variable like
>
> --> IF GLB > 2 GLB = 0
> dial(iax2/isp${GLB}/${EXTEN})
> --> GLB = GLB +1
>
> I believe this could cause a race condition if two calls are sent to the
> carrier at the same time?
Yes. If that's an issue for your carrier, use the decimal part of
${UNIQUEID}. It's incremented every time Asterisk creates a channel so
there would be no race condition.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] SIP provider registration attempts
Hi, I am registering my Asterisk boxes to a SIP provider for outgoing calls. My "outgoing" dialplan context tries to dial out in sequence, starting with the SIP provider then ISDN lines and finally analog lines. So the idea is that if the SIP trunk fails then all calls are dialed out via ISDN and analog. I noticed however that if I switch my DSL connection off (ie. no internet access to my SIP provider), Asterisk still tries to send calls out to the SIP provider and it doesn't fail over to the other trunks (at least not in an appropriate time lapse). When the DSL is down I get: "sip show registry": HostUsername Refresh StateReg. Time sip.provider.com:5060xx 105 Request Sent Tue, 23 Feb 2010 18:06:42 I've read about the sip registerattempts = Number option but: a) is this the option I'm really looking for? b) what happens if the DSL lines come back up? does Asterisk re-attempt registration automatically? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP provider registration attempts
Hi!
> My "outgoing" dialplan context tries to dial out in sequence, starting
> with the SIP provider then ISDN lines and finally analog lines.
> [...]
> When the DSL is down I get:
>
> "sip show registry":
>
> HostUsername Refresh State
> Reg. Time sip.provider.com:5060xx 105
> Request Sent Tue,
> 23 Feb 2010 18:06:42
Up front: Call routing has as such nothing to do with SIP registration.
Look at qualify= for sip.conf, and consider to extend your diaplan for a
better routing decision with a snippet like this:
exten => _00.,n,Set(VOIPCHECK=0)
exten => _00.,n,NoOp(-- ${PEERCHECK1} status:
${SIPPEER(${PEERCHECK1}:status)} --)
exten => _00.,n,ExecIf($["${SIPPEER(${PEERCHECK1}:status):0:2}" =
"OK"]|Set|VOIPCHECK=1)
where PEERCHECK1 corresponds to the provider section in sip.conf that you
would like to monitor.
Philipp
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Re: [asterisk-users] Denying call transfer to certain extensions
Ex-girlfriend is another answer to your query. Set it up like this Exten => 101,1,Verbose(let's call ext 101) Exten => 101/100,n,Dial(SIP/101,20,KkTt) Exten => 101/102,n,Dial(SIP/101,20,KkTt) Exten => 101,n,Playback(cant-dial-it) Exten => 101,n,hangup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Ossama Sent: Tuesday, February 23, 2010 5:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Denying call transfer to certain extensions But this won't help if 100 or 101 wants to call 102. What I want is, if a call coming from a trunk 100 rings, and if the caller wants to be transfered to 101, the transfer is denied. In other words, 101 can't get transfered calls. Danny Nicholas wrote: > Follow-me will most likely be your best bet for this trick. Say you have > extensions 100, 101 and 102. 100 is the receptionist, 101 is sales and 102 > is the boss, who doesn't want to be disturbed. If you set up followme on > 102 to go to voicemail or whatever, 102 won't ring. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Ossama > Sent: Monday, February 22, 2010 5:53 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Denying call transfer to certain extensions > > Hi all, > > Is there a way to deny call transfers to certain extensions? > > Thanks, > Ahmed Ossama > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP provider registration attempts
--- On Tue, 2/23/10, Philipp von Klitzing
wrote:
> Look at qualify= for sip.conf, and consider to extend your
> diaplan for a
> better routing decision with a snippet like this:
>
> exten => _00.,n,Set(VOIPCHECK=0)
> exten => _00.,n,NoOp(-- ${PEERCHECK1} status:
> ${SIPPEER(${PEERCHECK1}:status)} --)
> exten =>
> _00.,n,ExecIf($["${SIPPEER(${PEERCHECK1}:status):0:2}" =
> "OK"]|Set|VOIPCHECK=1)
>
> where PEERCHECK1 corresponds to the provider section in
> sip.conf that you
> would like to monitor.
Thanks Philipp!
Do you suggest I use qualify=yes (default 1 or 2 seconds?) or should I specify
a value in ms?
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Re: [asterisk-users] Calls per second limit in manager
Also, why are you saying your name is Philip? On 24/02/2010, at 12:59 AM, CDR wrote: > My dear friend Matt Riddell insists that the Manager only can dial 5 > calls per seconds, which I find ridiculous. Is there a way to prove > him wrong and have him lift the limit that has been plaguing the > life of us users of SineDialer and SmoothTorrque > Philip > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
The responses from the Asterisk manager on your machine start providing responses of no account code when calls are initiated at a higher rate. On 24/02/2010, at 12:59 AM, CDR wrote: > My dear friend Matt Riddell insists that the Manager only can dial 5 > calls per seconds, which I find ridiculous. Is there a way to prove > him wrong and have him lift the limit that has been plaguing the > life of us users of SineDialer and SmoothTorrque > Philip > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which H.323 to use in Ast 1.6
We're doing a project that requires H.323 to an Avaya. Does anyone have experience to share on which H.323 driver to use in asterisk 1.6? Is the diference between h323 and ooh323 still worth the extra effort? (We've only installed h323 under 1.4) If you have setup/config experience with this setup in Asterisk 1.6 please share! Thanks, MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX devices not registering after upgrade to
>> Hi All, >> We have encountering issue that IAX enable voice gateways not registering with asterisk after upgrade from asterisk 1.4.18.1 -> 1.4.29 >> Before that IAX works very well. >> If any one have similar issue and solution for that let me know. Check for ERROR[] chan_iax2.c: Call rejected, CallToken Support required. If add requirecalltoken=no to Iax trunk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
So you're saying that you could at least theoretically push more than 5 CPS through, you just would get a lot of "no account code" responses? Reading the SmoothTorrque Wiki, I could see where a user might want to process more than 300 CPM (5*60), but if I'm going to spend the money for over 300 phone lines, I'm probably going to use either Commercial Asterisk or another application. Am I barking up the wrong tree? Regards, Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Tuesday, February 23, 2010 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls per second limit in manager The responses from the Asterisk manager on your machine start providing responses of no account code when calls are initiated at a higher rate. On 24/02/2010, at 12:59 AM, CDR wrote: > My dear friend Matt Riddell insists that the Manager only can dial 5 > calls per seconds, which I find ridiculous. Is there a way to prove > him wrong and have him lift the limit that has been plaguing the > life of us users of SineDialer and SmoothTorrque > Philip > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running safe_asterisk
>> On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote: >> About two weeks ago there was a thread about asterisk suddenly dying >> - I posted a response that the same happens to my asterisk about once >> a month, sometimes more. >> Someone suggested using 'safe_asterisk' (and get hold of a core dump) >> which sounds like a good idea, but one thing I can't figure is how to >> get "module reload app_queue" executed automatically at startup? > >> All modules are loaded at startup. Why would you need a reload? > >To be honest I don't remember any more, I just know my queueing doesn't work unless I reload. I think it's a timing issue at startup - that app_queue gets loaded too early or something. ah, >here is my question about the same, but back in 2007: >http://lists.digium.com/pipermail/asterisk-users/2007-May/188072.html Add to below to end of modules.conf, your app_queue is loading before your extensions are available causing static agents to say (Invalid) when you do a show queue. A normal "reload" reloads modules in correct sequence. This resolves permanently. load => pbx_config.so load => chan_local.so load => chan_sip.so load => app_queue.so Rudi Oosthuizen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
23 feb 2010 kl. 20.18 skrev Matt Riddell: > The responses from the Asterisk manager on your machine start > providing responses of no account code when calls are initiated at a > higher rate. > Where's the bug report id? I haven't heard about this limit. I don't know what it is, but we should at least be able to accept the originate requests in asynch mode, put them on a queue and process them in a separate thread (which can be configurable in manager.conf). This is just brainstorming - but first, let's try to find out if the limit you believe in exists in the code or is just the effect of something else. /O > > > On 24/02/2010, at 12:59 AM, CDR wrote: > >> My dear friend Matt Riddell insists that the Manager only can dial 5 >> calls per seconds, which I find ridiculous. Is there a way to prove >> him wrong and have him lift the limit that has been plaguing the >> life of us users of SineDialer and SmoothTorrque >> Philip >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - o...@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Olle E. Johansson skrev: > 23 feb 2010 kl. 20.18 skrev Matt Riddell: > >> The responses from the Asterisk manager on your machine start >> providing responses of no account code when calls are initiated at a >> higher rate. >> > Where's the bug report id? > > I haven't heard about this limit. I don't know what it is, but we should at > least be able to accept the originate requests > in asynch mode, put them on a queue and process them in a separate thread > (which can be configurable > in manager.conf). This is just brainstorming - but first, let's try to find > out if the limit > you believe in exists in the code or is just the effect of something else. > It seems to be the effect of something else - or perhaps an older asterisk version. I wrote a quick script that ran 80 calls in ~ 0.2 seconds with no problems what so ever. I am using asterisk 1.6.2.1, and the script authenticated for each time, and used the originate-application for it's calls. - - T >> >> On 24/02/2010, at 12:59 AM, CDR wrote: >> >>> My dear friend Matt Riddell insists that the Manager only can dial 5 >>> calls per seconds, which I find ridiculous. Is there a way to prove >>> him wrong and have him lift the limit that has been plaguing the >>> life of us users of SineDialer and SmoothTorrque >>> Philip >>> -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuEO2EACgkQ573V05EH/pZDQwCfaotZoweNLI8cTQ+yxZ2tr7WK +YsAn3OxXc5ULAj4lPdiIhoBDG4Tm7Xp =XE12 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
Yeah, the problem's not the origination. The problem is that calls originated asyn with accountcodes show up in show channels concise without details. Pretty simple to test with sipp and core show channels concise. I assume it's because the call origination happens at a faster rate than Asterisk can fill out the details. Apologies for top post, laptop is running a defrag. On 24/02/2010, at 9:32 AM, Tommy Botten Jensen wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA512 > > Olle E. Johansson skrev: >> 23 feb 2010 kl. 20.18 skrev Matt Riddell: >> >>> The responses from the Asterisk manager on your machine start >>> providing responses of no account code when calls are initiated at a >>> higher rate. >>> >> Where's the bug report id? >> >> I haven't heard about this limit. I don't know what it is, but we >> should at least be able to accept the originate requests >> in asynch mode, put them on a queue and process them in a separate >> thread (which can be configurable >> in manager.conf). This is just brainstorming - but first, let's try >> to find out if the limit >> you believe in exists in the code or is just the effect of >> something else. >> > > It seems to be the effect of something else - or perhaps an older > asterisk version. I wrote a quick script that ran 80 calls in ~ 0.2 > seconds with no problems what so ever. > > I am using asterisk 1.6.2.1, and the script authenticated for each > time, > and used the originate-application for it's calls. > > - - T > >>> >>> On 24/02/2010, at 12:59 AM, CDR wrote: >>> My dear friend Matt Riddell insists that the Manager only can dial 5 calls per seconds, which I find ridiculous. Is there a way to prove him wrong and have him lift the limit that has been plaguing the life of us users of SineDialer and SmoothTorrque Philip -- > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.9 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iEYEAREKAAYFAkuEO2EACgkQ573V05EH/pZDQwCfaotZoweNLI8cTQ+yxZ2tr7WK > +YsAn3OxXc5ULAj4lPdiIhoBDG4Tm7Xp > =XE12 > -END PGP SIGNATURE- > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
-BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Matt Riddell skrev: > Yeah, the problem's not the origination. > > The problem is that calls originated asyn with accountcodes show up in > show channels concise without details. > > Pretty simple to test with sipp and core show channels concise. > > I assume it's because the call origination happens at a faster rate > than Asterisk can fill out the details. > > Apologies for top post, laptop is running a defrag. > No worries. Did I misunderstand the bit about this being calls spawned from the AMI (Manager) only? And what details are missing? 'sip show concise' gives me the following: SIP/05-00ca!internal!!1!Ringing!(None)!!100!!3!26!(None)!1266961653.567 ... and similar * n. Thanks, Tommy -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuETagACgkQ573V05EH/pb73QCgnUMlaghEOBjS9aVw5PKtJZy4 Tl0AoJ2kRTHr0tOJPxsXAdVPmoulGJJE =zS/n -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
Yeah, so at say 10 calls per second originated from the manager with async on, you'd likely have about a thousand channels. Then if you type show channels concise you'll see about 20% of the calls are missing accountcode, destination etc. I wrote some code to just repeat this test over and over, and with 5 CPS you get maybe one or two channels in this state, but as you increase the CPS you end up with more. Initially I thought it might have been manager parsing, but did show channels concise from Asterisk console and got the same. On 24/02/2010, at 10:50 AM, Tommy Botten Jensen wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA512 > > Matt Riddell skrev: >> Yeah, the problem's not the origination. >> >> The problem is that calls originated asyn with accountcodes show up >> in >> show channels concise without details. >> >> Pretty simple to test with sipp and core show channels concise. >> >> I assume it's because the call origination happens at a faster rate >> than Asterisk can fill out the details. >> >> Apologies for top post, laptop is running a defrag. >> > No worries. > > Did I misunderstand the bit about this being calls spawned from the > AMI > (Manager) only? > > And what details are missing? > 'sip show concise' gives me the following: > SIP/05-00ca!internal!!1!Ringing!(None)!!100!!3!26!(None)! > 1266961653.567 > ... and similar * n. > > > Thanks, > > Tommy > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.9 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iEYEAREKAAYFAkuETagACgkQ573V05EH/pb73QCgnUMlaghEOBjS9aVw5PKtJZy4 > Tl0AoJ2kRTHr0tOJPxsXAdVPmoulGJJE > =zS/n > -END PGP SIGNATURE- > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] directrtp with SIP + H.323
On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis wrote: > We're creating a SIP gateway for a client that will take one leg of a call > in via SIP, and out the other side via H.323. To minimize load on the > gateway, we would like to have the RTP stream bypass the gatewayy altogether > (directrtp/reinvite). Is this possible with these to protocols? > > Thanks Yate claims it can do this: http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macros, GoSub & StackPop
Hi - I have a Macro that contains a GoTo. The documentation indicates: If you GoTo out of the Macro context, the Macro will terminate and control will return at the location refered to by the Goto. I thought I might convert the Macro to a GoSub routine, but the documentation doesn't mention what happens if you GoTo out. It does however mention that the return address gets pushed onto the stack, so I'm a little concerned about the state of the stack if I simply GoTo out. Should I call StackPop first? Thanks! Hugh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macros, GoSub & StackPop
On Tuesday 23 February 2010 21:35:39 hugolivude wrote: > Hi - > > I have a Macro that contains a GoTo. The documentation indicates: > > If you GoTo out of the Macro context, the Macro will terminate and control > will return at the location refered to by the Goto. > > I thought I might convert the Macro to a GoSub routine, but the > documentation doesn't mention what happens if you GoTo out. It does > however mention that the return address gets pushed onto the stack, so I'm > a little concerned about the state of the stack if I simply GoTo out. > Should I call StackPop first? Yes, that is the purpose of the StackPop function: to remove the last stack frame without returning to the Gosub location. Note that if you had any LOCAL() variables (or arguments), those will additionally all be cleared with the execution of the StackPop application (and the previous values, if any, will return). -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX devices not registering after upgrade to asterisk
> > Message: 18 > Date: Tue, 23 Feb 2010 15:02:24 +0100 > From: Philipp von Klitzing > Subject: Re: [asterisk-users] IAX devices not registering after >upgrade to asterisk 1.4.29 > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: ><4b83ee00.10686.84d...@klitzing.pool.informatik.rwth-aachen.de> > Content-Type: text/plain; charset=US-ASCII > Hi philip, > The Issue sorted. thanks for sharing the details. Regards, Vidura. > Hi! > > > We have encountering issue that IAX enable voice gateways not > > registering with asterisk after upgrade from asterisk 1.4.18.1 -> 1.4.29 > > Before that IAX works very well. If any one havesimilarissue and solution > > for that let me know. > > Search or google for "calltokenoptional". > > http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf > http://downloads.asterisk.org/pub/security/AST-2009-006.html > > Philipp > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Denying call transfer to certain extensions
On Tue, Feb 23, 2010 at 7:50 PM, Danny Nicholas wrote: > What I want is, if a call coming from a trunk 100 rings, and if the > caller wants to be transfered to 101, the transfer is denied. In other > words, 101 can't get transfered calls. WHat about using featuresmap to replace the usual transfer application with code that tests to see the origin of the cal ind if it is from the 100 do something else, otherwise transfer as expected. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
