Re: [asterisk-users] Asterisk NOT in the media path
On 02/24/2012 10:51 PM, Jared Geiger wrote:
On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls
to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call,
and based upon
some checks in the dialplan send the call through to one
of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to
stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes
in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1
when the call
has been set up.
As long as there are no NATs involved, yes, this should work.
You will also need 'canreinvite' ('directmedia' in Asterisk
1.8 and later) in the peer definition for the provider.
Hello again,
this is currently not really working.
I see on the Asterisk CLI that the call streams through my
Asterisk A1 (which should stay out of the media path) :
[Feb 23 22:24:47] -- Called Mast/980419
[Feb 23 22:24:47] -- SIP/Mast-000e answered
SIP/VOXBONEin-000d
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-000d
and SIP/Mast-000e
*CLI
*CLI core show channels
Channel Location State Application(Data)
SIP/Mast-00 (None) Up AppDial((Outgoing Line))
SIP/VOXBONEin-00 980419@VOXBONEin Up Dial(SIP/Mast/980419)
2 active channels
1 active call
Peer VoxBone and peer Mast should re-invite and leave this
Asterisk out of the media path on call answer.
These are my SIP peer definitions :
[VOXBONEin]
type=peer
host=XX.XX.XX.XX
context=VOXBONEin
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
[Mast]
type=peer
host=XX.XX.XX.XX
defaultuser=Mast
secret=guessme
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
Am I missing a setting ? Using Asterisk 1.6.2.22
The Asterisk server still stays in the SIP Signaling path of the call,
just media does not flow through the server. You can verify this by
running a SIP debug and looking at the media endpoints.
What is it that I should be looking for in the SIP debug information ?
Is it in the SDP-body ?
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NOT in the media path
On Feb 29, 2012, at 7:25 AM, Jonas Kellens wrote: The Asterisk server still stays in the SIP Signaling path of the call, just media does not flow through the server. You can verify this by running a SIP debug and looking at the media endpoints. What is it that I should be looking for in the SIP debug information ? Is it in the SDP-body ? To set up direct media, Asterisk will send a re-invite with an SDP body containing the address of the other endpoint. RTP then flows directly between endpoints. If you just want to know if Asterisk is in the media path or not, you can also use rtp set debug ... and Asterisk will log a line for each RTP packet it handles. -- v: 248.893.0738 | f: 248.893.0747 http://macprofessionals.com/ find us on facebookall -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:28 PM, Alejandro Imass a...@p2ee.org wrote: What you are saying seems impossible and makes no sense unless the router is assigning a public IP or is SIP aware and knows how to read the routing data contained inside the SIP packets, and none of the consumer routers are SIP aware AFAIK, especially not the WRT-54G. You can: a. Continue to tell us that what we are doing every day is impossible. or... b. Go set up a test in your lab with an Asterisk server, a cheap router, and a SIP client and see. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just work. We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. Our set-up is fundamentally public and private Asterisk servers running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD 8.2 and Asterisk 1.8 but we are in that process right now. Some Asterisk run in jails so I can understand the NAT issues there may be caused by the server itself. I honestly *love* your OpenVPN idea but I have to find a cheap ATA that could run as an OpenVPN client. Taking the simplest example a simple Asterisk 1.6 server on a public IP running on the base system (not in a jail): We run an operation that spans several countries including Canada, the US and the Latin American Andean region. As examples, with Canadian ISPs such as Rogers and Bell we have always had to redirect the ports and use STUN server for the HT-286 to register to the Asterisk server. In the US we have the same problem with Comcast networks, so I don't understand how you say that you plug a Grandtream SIP ATA to a Comcast router and it just works. However, in a couple of NOLA countries the ISP's routers actually give public IPs, so if the SIP ATAs are connected directly to the ISP router, or in the DMZ then it just works as you say, BUT if the ATA is connected behind the firewall, or to a WiFi router, then we've _allways_ had to redirect ports. In every sigle customer we have had to send instructions on how to redirect ports, and of course to configure firewall if present. I just don't understand how you and other here say that a SIP ATA can just work. On the contrarty, with IAX2 using cheap AG-188N from Atcom they are just plug and play when shipped with a standard conf, and we have none of the quality issues you are referring to. We do have some call drops however, and some hangup problems but they don't affect our clients as much as having to deal with NAT issues. We may not run 15K extensions like you but I think we have a pretty good testing ground and have dealt with a fair share of NAT problems with SIP, that you and others here apparently don't have, so I am as amazed by your likeness of SIP as perhaps you are amazed as our likeness of IAX. -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On 02/29/2012 08:22 AM, Alejandro Imass wrote: We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting Ulimit Message after restart asterisk service
Hi all, Currently I'm getting this message after restarting asterisk service; Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted Before when I had root access I was not facing this message after that system administrator assigned me sudo access for restarting asterisk service. Please assist me out to resolve this issue at earliest. I also tried to set ulimit till 4-40 times higher than currently set i.e. 1024 but still giving me same message. -- Regards, Ahmed Munir Chohan -- _ -- 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] Getting Ulimit Message after restart asterisk service
Hi all, Currently I'm getting this message after restarting asterisk service; Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted Before when I had root access I was not facing this message after that system administrator assigned me sudo access for restarting asterisk service. Please assist me out to resolve this issue at earliest. I also tried to set ulimit till 4-40 times higher than currently set i.e. 1024 but still giving me same message. -- Regards, Ahmed Munir Chohan -- _ -- 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] Getting Ulimit Message after restart asterisk service
This one is simple. Open /usr/sbin/safe_asterisk and put # in first character of line 86 and 102. Or modfy /etc/sudoers to allow your sudo to execute ulimit. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, February 29, 2012 8:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Ulimit Message after restart asterisk service Hi all, Currently I'm getting this message after restarting asterisk service; Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted Before when I had root access I was not facing this message after that system administrator assigned me sudo access for restarting asterisk service. Please assist me out to resolve this issue at earliest. I also tried to set ulimit till 4-40 times higher than currently set i.e. 1024 but still giving me same message. -- Regards, Ahmed Munir Chohan -- _ -- 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] 10.2.0-rc2: permitted contact can't register.
On 02/29/2012 02:30 AM, Zohair Raza wrote: You want to allow single IP or whole subnet ? Regards, Zohair Raza On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com mailto:seandar...@gmail.com wrote: An outside device can't register: WARNING: getnameinfo(): ai_family not supported WARNING: chan_sip.c:14456 parse_register_contact: Domain '69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP ) sip.conf: [general] ... alwaysreject=yes dynamic_exclude_static = yes allowguest=no contactdeny=0.0.0.0/0.0.0.0 http://0.0.0.0/0.0.0.0 contactpermit=69.0.0.0/255.0.__0.0 http://69.0.0.0/255.0.0.0 I've also tried without any contactdeny. Same result. I'm completely puzzled. Any help appreciated. sean I don't care. Single ip, subnet, or nothing at all, nothing allows registration. sean -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 02/29/2012 08:22 AM, Alejandro Imass wrote: [...] The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. My point of view is that we've had many problems with SIP and NAT and that IAX just works great for us, and that in *our* experience IAX has worked better for us. Just to clear my head up a bit: are you supporting the argument that SIP is better for Asterisk than IAX? -- _ -- 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] 10.2.0-rc2: permitted contact can't register.
On 12-02-29 10:15 AM, sean darcy wrote: On 02/29/2012 02:30 AM, Zohair Raza wrote: You want to allow single IP or whole subnet ? Regards, Zohair Raza On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com mailto:seandar...@gmail.com wrote: An outside device can't register: WARNING: getnameinfo(): ai_family not supported WARNING: chan_sip.c:14456 parse_register_contact: Domain '69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP ) sip.conf: [general] ... alwaysreject=yes dynamic_exclude_static = yes allowguest=no contactdeny=0.0.0.0/0.0.0.0 http://0.0.0.0/0.0.0.0 contactpermit=69.0.0.0/255.0.__0.0 http://69.0.0.0/255.0.0.0 I've also tried without any contactdeny. Same result. I'm completely puzzled. Any help appreciated. sean I don't care. Single ip, subnet, or nothing at all, nothing allows registration. Showing us the problem would be helpful too, please attach a debug[1] log. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check 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
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 02/29/2012 09:25 AM, Alejandro Imass wrote: On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Flemingkpflem...@digium.com wrote: On 02/29/2012 08:22 AM, Alejandro Imass wrote: [...] The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. My point of view is that we've had many problems with SIP and NAT and that IAX just works great for us, and that in *our* experience IAX has worked better for us. Just to clear my head up a bit: are you supporting the argument that SIP is better for Asterisk than IAX? I have no idea where you got that sort of conclusion. I was making a statement to counter your repeated arguments that using SIP behind a NAT without special configuration is 'impossible'. It's clearly not impossible, it's not even impractical. It is commonplace. Certainly there are plenty of examples of SIP endpoints working poorly behind NAT devices, and replacing that endpoint with an IAX2 endpoint curing the symptoms. Invariably, this is caused by the fact that the NAT device was attempting to 'help' the SIP endpoint, and failed miserably. In every case I can remember, turning off any SIP-specific functionality in that NAT device (which is not always possible) allowed the SIP endpoint to work as expected. There are certainly scenarios where deploying a SIP endpoint behind a NAT can be problematic; usually, these revolve around deploying a SIP *server* behind a NAT, but even this can be handled reasonably well by configuration options already present in Asterisk. Deploying SIP *clients* behind NATs, talking to a SIP server that is on a public IP, is generally trivial and takes no special effort at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass a...@p2ee.org wrote: On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] Yes, I have had no problems with Grandstream first gen ATAs, configured with server and credentials and shipped off, they just work. We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. Our set-up is fundamentally public and private Asterisk servers running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD 8.2 and Asterisk 1.8 but we are in that process right now. Some Asterisk run in jails so I can understand the NAT issues there may be caused by the server itself. I honestly *love* your OpenVPN idea but I have to find a cheap ATA that could run as an OpenVPN client. Taking the simplest example a simple Asterisk 1.6 server on a public IP running on the base system (not in a jail): We run an operation that spans several countries including Canada, the US and the Latin American Andean region. As examples, with Canadian ISPs such as Rogers and Bell we have always had to redirect the ports and use STUN server for the HT-286 to register to the Asterisk server. In the US we have the same problem with Comcast networks, so I don't understand how you say that you plug a Grandtream SIP ATA to a Comcast router and it just works. However, in a couple of NOLA countries the ISP's routers actually give public IPs, so if the SIP ATAs are connected directly to the ISP router, or in the DMZ then it just works as you say, BUT if the ATA is connected behind the firewall, or to a WiFi router, then we've _allways_ had to redirect ports. In every sigle customer we have had to send instructions on how to redirect ports, and of course to configure firewall if present. I just don't understand how you and other here say that a SIP ATA can just work. On the contrarty, with IAX2 using cheap AG-188N from Atcom they are just plug and play when shipped with a standard conf, and we have none of the quality issues you are referring to. We do have some call drops however, and some hangup problems but they don't affect our clients as much as having to deal with NAT issues. We may not run 15K extensions like you but I think we have a pretty good testing ground and have dealt with a fair share of NAT problems with SIP, that you and others here apparently don't have, so I am as amazed by your likeness of SIP as perhaps you are amazed as our likeness of IAX. If you can post some SIP debug info from an ATA trying to register without any redirection and also the relevant portions of your sip.conf, I am sure I can help. Do it from a new location with an el cheapo home router, Linksys WRTXXX. If I cannot help you in a few emails, we can take this offline. Actually paste your entire sip.conf in pastebin or something, as well as sip debug. Also the configs of your ATAs. I think you have over-engineered to the point of creating problems. This is very common. My philosophy is KISS Thanks, Steve T -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 8:34 AM, Kevin P. Fleming kpflem...@digium.com wrote: Certainly there are plenty of examples of SIP endpoints working poorly behind NAT devices, and replacing that endpoint with an IAX2 endpoint curing the symptoms. Invariably, this is caused by the fact that the NAT device was attempting to 'help' the SIP endpoint, and failed miserably. In every case I can remember, turning off any SIP-specific functionality in that NAT device (which is not always possible) allowed the SIP endpoint to work as expected. We have *never* found a single SIP helper to actually help anything. They always break everything. The only SIP-related setting I can think of that works is Enable consistent NAT found in Sonicwall routers (but turn off all other SIP helpers). The WRT54G and Apple Airport come to mind as stable and reliable home/small business routers that seem to just work. The WRTs seem to go bad every few years and need an occasional reboot, the Airports work forever in our experience without being touched. All stock out of the box. options already present in Asterisk. Deploying SIP *clients* behind NATs, talking to a SIP server that is on a public IP, is generally trivial and takes no special effort at all. This is today's reality. That's why I mentioned the possibility that he's using ancient devices and routers, or that Yugo. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/29/2012 08:22 AM, Alejandro Imass wrote: We use the HT-286, the server is on a public IP the nat setting on asterisk is set to yes and without port re-direction the ATAs have never connected from a private network, so I honestly find this SIP plug and play very hard to believe. But if it is true, then maybe you can actually help us figure out all the NAT issues we've had with SIP for the past 5 years. Perhaps, it is simply ignorance on our side and we have something fundamentally wrong in our set-up somewhere that may be have been causing these issues with NAT. The number of 'plain' SIP endpoints deployed behind consumer-grade NAT devices talking to Asterisk servers on public IP addresses is in the millions, if not the tens of millions. As has already been posted, Asterisk itself handles all the far-end NAT traversal duties necessary for this to work; neither the remote endpoint nor the NAT device need to do anything special, nor do they require any configuration. Rather than post a lengthy exposition on how widespread your network is and how technically astute your people are, you would probably accomplish much more to setup a simple test scenario as has been previously suggested, and if it does not work for you, post the details of the scenario and the failure here. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org Agreed with one exception, the endpoint behind the NAT DOES need to be setup correctly to keep the router from seeing inbound traffic to the device as unsolicited and drop it. That is a function of the router but keep alives from Qualify on the Asterisk side, and setting the device to register every few minutes will keep that mapping open and alive, letting traffic pass as solicited. Thanks, Steve Totaro -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Agreed with one exception, the endpoint behind the NAT DOES need to be setup correctly to keep the router from seeing inbound traffic to the device as unsolicited and drop it. That is a function of the router but keep alives from Qualify on the Asterisk side, and setting the device to register every few minutes will keep that mapping open and alive, letting traffic pass as solicited. We use qualify=yes on Asterisk and a few months ago turned OFF the keep-alive feature on all SIP clients on our entire system. This is working fine, and we did it because of a strange bug/behavior with certain versions of Cisco SPA series firmware. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez car...@televolve.comwrote: On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: Agreed with one exception, the endpoint behind the NAT DOES need to be setup correctly to keep the router from seeing inbound traffic to the device as unsolicited and drop it. That is a function of the router but keep alives from Qualify on the Asterisk side, and setting the device to register every few minutes will keep that mapping open and alive, letting traffic pass as solicited. We use qualify=yes on Asterisk and a few months ago turned OFF the keep-alive feature on all SIP clients on our entire system. This is working fine, and we did it because of a strange bug/behavior with certain versions of Cisco SPA series firmware. -- Carlos Alvarez TelEvolve 602-889-3003 So you turned it off on the phones but use it on the Asterisk side? Do you set a value or just use qualify=yes? I had many problems with qualify over VSAT as ping times and jitter are crazy. 700ms ping times were considered Good from the IZ in Iraq to Equinix data center in VA, it took some tweaking to find the right value so a phone that was Reachable was not labeled Unreachable, I did want phones that were truly unreachable to be marked as such, more to spot patterns and act on them or with the vendor. Did you submit a bug report? If it is easy to reproduce and you feel like helping out, report it. I do not report issues if there is a simple way to do the same thing, but I know I should. What does the debug or strange behavior look like? Probably a variance in the RFC implementation. Thanks, Steve T -- _ -- 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] 10.2.0-rc2: permitted contact can't register.
An outside device can't register: WARNING: getnameinfo(): ai_family not supported WARNING: chan_sip.c:14456 parse_register_contact: Domain '69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP ) sip.conf: [general] ... alwaysreject=yes dynamic_exclude_static = yes allowguest=no contactdeny=0.0.0.0/0.0.0.0 contactpermit=69.0.0.0/255.0.0.0 I tried reproducing this with 10.2.0-rc3 (which contains only three additional patches with respect to 10.2.0-rc2, none of which have anything to do with ACLs), and was unable to. A snippet of my configuration: [general] context=blah allowguest=no udpbindaddr=0.0.0.0 contactdeny=0.0.0.0/0.0.0.0 contactpermit=10.0.0.0/255.0.0.0 [7001](natted-phone,polycom_430) context = blah username= 7001 language= en secret = blahblahblah qualify = yes [7003](natted-phone,polycom_430) context = blah username= 7003 language= en secret = blahblahblah qualify = yes Peer 7001 is registers from subnet 10.x.y.z, while peer 7003 registers from 10.x.p.q. With the above configuration, both peers can register. With the following configuration change: contactdeny=0.0.0.0/0.0.0.0 contactpermit=10.x.y.0/255.255.255.0 Peer 7003's registration is rejected, as expected. So, I'm not sure what the configuration issue is - you may need to either post the rest of your configuration, or provide - as Paul suggested - either the actual SIP REGISTER request or some a portion of the DEBUG log. I've also tried without any contactdeny. Same result. I'm completely puzzled. Any help appreciated. sean Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check 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
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 8:58 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: So you turned it off on the phones but use it on the Asterisk side? Do you set a value or just use qualify=yes? Yes, just as I said, just qualify=yes. Did you submit a bug report? If it is easy to reproduce and you feel like helping out, report it. I do not report issues if there is a simple way to do the same thing, but I know I should. Cisco makes it too difficult to submit bugs so I just don't care to help them. When we find a service-impacting bug we report it to our distributor, who tests it and presumably reports it, but I'm not sure. Also I wasn't sure if it was an Asterisk bug or Cisco bug, and didn't care enough to find out since a clean work-around was possible. What does the debug or strange behavior look like? Probably a variance in the RFC implementation. Soon as the keep-alive packet is sent, the phone is no longer reachable from Asterisk. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2
Hi, while testing asterisk 1.8.10-rc2 I stumbled across a weird behavior. I want to notify a snom phone to reload its configuration. For this to happen, I use the NOTIFY mechanism. I started the notify via AMI command. Asterisk is bound to udp 25060, because all phones are registered with a local opensips proxy which uses 5060. The expected behavior would be: asterisk send SIP NOTIFY to the proxy, the proxy sends it to the phone. Actually asterisk sends the packet to the proxy, but the contact header contains something invalid (IMHO): On Manager Interface: T 127.0.0.1:57530 - 127.0.0.1:5038 [AP] Action: SIPnotify. Channel: SIP/max. Variable: Event=check-sync\;reboot=false. Leads to: U 192.168.10.72:25060 - 192.168.10.72:5060 NOTIFY sip:max@192.168.10.72 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.72:0;branch=z9hG4bK1dff6efe. Max-Forwards: 70. From: asterisk sip:asterisk@192.168.10.72;tag=as66766c2a. To: sip:max@192.168.10.72. Contact: sip:asterisk@192.168.10.72:0. Call-ID: 412a8eff76bd7ac56ac06831256fd6aa@192.168.10.72. CSeq: 102 NOTIFY. Subscription-State: terminated. Event: check-sync;reboot=false. Content-Length: 0. The weird thing is the port number 0 in the contact header. Is this a bug or do I something wrong? Thanks, Karsten -- _ -- 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] Problem while sending SIP NOTIFY via AMI in 1.8.10-rc2
Hi, a little extension to my previous post: The phone sends 200 OK for the NOTIFY via proxy to asterisk, but asterisk seems to ignore this. About 500 ms later, the NOTIFY is repeated by asterisk. This continues up to the final timeout (with the typical log message). Karsten -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-28 21:22:44 +, Kevin P. Fleming said: On 02/28/2012 03:08 PM, Troy Telford wrote: [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. trunk=yes was the source of the problem. So now I suppose I'll have trunk=no while I patiently wait for the fix to appear in Debian. - As an aside: I'm perfectly capable of compiling Asterisk; I prefer to use the packages for pretty much all of the reasons packages were invented. -- Troy Telford -- _ -- 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 distributions
I'm looking at replacing a PBX for a small business with an asterisk box. I'm rather attracted to the idea of one of the iso distributions where someone did most of the integration for us already ;) Can anyone comment on the pros/cons of the various options? I'm seeing several options out there: -Trixbox CE (no new version since 2010? is this project dead?) -Asterisk NOW -PBX in a Flash -Elastix -- _ -- 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 distributions
Asterisk Now should serve your needs nicely. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett Sent: Wednesday, February 29, 2012 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk distributions I'm looking at replacing a PBX for a small business with an asterisk box. I'm rather attracted to the idea of one of the iso distributions where someone did most of the integration for us already ;) Can anyone comment on the pros/cons of the various options? I'm seeing several options out there: -Trixbox CE (no new version since 2010? is this project dead?) -Asterisk NOW -PBX in a Flash -Elastix -- _ -- 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] asterisk distributions
FreePBX have also an ISO distribution - I would recommend to use that one. HTH, Ioan On Wed, Feb 29, 2012 at 7:43 PM, Danny Nicholas da...@debsinc.com wrote: Asterisk Now should serve your needs nicely. -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On 02/29/2012 11:35 AM, Troy Telford wrote: On 2012-02-28 21:22:44 +, Kevin P. Fleming said: On 02/28/2012 03:08 PM, Troy Telford wrote: [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely. trunk=yes was the source of the problem. Thanks for following up! The patch to resolve this problem was very small, but I understand your desire to wait for a package. It will be in the 1.8.11 release, although of course the Debian team could choose to backport the one-line fix into their existing release if they so choose. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great
On 2012-02-29 15:25:49 +, Alejandro Imass said: We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. The original question (mine) was that my sound quality when using IAX was bad; with SIP the sound quality was great. Critically, I mentioned that I wanted to use IAX; I even said I was willing to do some self torture to get IAX working properly. I only wanted some help in figuring out what was 'wrong' with my IAX configuration. After a few suggestions, Kevin Fleming noticed I was using trunk=yes, and it was likely that my Asterisk install was being affected by a just-fixed bug. Disabling trunking fixed the problem - the voice sounds great even in my worst-case scenerio (which was always almost unintelligible). The devolution into a flamewar is unfortunate, but such things are inevitable whenever a 'this' vs 'that' question is posed. For instance, is the Yugo really any worse than the competing Trabant? The only correct answer is to fling them both with a Trebuchet and see which one flies farther. -- Troy Telford -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 10:34 AM, Steve Totaro stot...@asteriskhelpdesk.com wrote: [...] If you can post some SIP debug info from an ATA trying to register without any redirection and also the relevant portions of your sip.conf, I am sure I can help. Do it from a new location with an el cheapo home router, Linksys WRTXXX. Yeah, I think it's time for me to shut up about SIP/NAT problems and, like you Carlos and Kevin pointed out, run a clean un-contaminated test lab to see if we can determine why our current set-up is so problematic with SIP and NAT. If I cannot help you in a few emails, we can take this offline. Thanks for offering to help. I will set-up a test lab but it's gonna take me some time to free a public server to do so. But it is obvious that the problem is on our side after reading all the responses. After all, VoIP is *not* by any means our core business we just use it as a tool, and up until now I thought that *everyone* using SIP ATAs and Asterisk had these NAT woes, so we just assumed it was so, and thought that mostly everyone had to perform particular configurations on the endpoints. It now seems obvious we are wrong. Anyway, my whole argumentative line in this thread is that in our particular case we found that IAX2 works great for _our_ set-ups and we don't share the view that IAX2 is a broken bat, and that in fact for us it just works great. Thanks, -- Alejandro Imass -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-29 15:25:49 +, Alejandro Imass said: We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. The original question (mine) was that my sound quality when using IAX was bad; with SIP the sound quality was great. Critically, I mentioned that I wanted to use IAX; I even said I was willing to do some self torture to get IAX working properly. Yeah, I wasn't referring particularly to the original post, just the way the thread turned against IAX like if it's not a viable solution and my point all along has been that for *us* IAX2 endpoints have worked better and easier to configure than SIP ones. Then it turned into a pissing contest, like you say, it happens in every list with the topic this or that. Again, as I pointed out to Steve above, and after reading all of your responses, our SIP/NAT woes seem obviously ignorance on our part, but that doesn't shadow the fact that IAX2 is working great for us with el-cheapo endpoints like Atcom's AG-188N and I would wish that many more manufacturers supported IAX2. We are happy with IAX and honestly never even had the need/curiosity to deal with the many SIP/NAT problems where sometimes it works great, and other times is a real pain in the ass that takes huge amounts of support to fix, and unhappy customers. On the other hand, IAX took some engineering efforts at first, but the support issues are practically non-existent. -- Alejandro Imass -- _ -- 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] GSM gateway call redirect
Hi, I have a Portech mv-374 GSM IP gateway and I have to redirect all the incoming calls to a certain phone number on every weeknight and all the weekend. What would be the best solution? I have to do it by asterisk because I have to record all the communication. Thanks for your help in advance. Best regards, Peter Gelencser -- _ -- 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] Same provider - IAX sounds bad, SIP sounds great
On Wed, Feb 29, 2012 at 1:26 PM, Alejandro Imass a...@p2ee.org wrote: On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote: On 2012-02-29 15:25:49 +, Alejandro Imass said: We use SIP and IAX interchangeably, but had less hassle with IAX. The topic of the discussion on this thread was that SIP is so awesome and that IAX is a peice of crap. The original question (mine) was that my sound quality when using IAX was bad; with SIP the sound quality was great. Critically, I mentioned that I wanted to use IAX; I even said I was willing to do some self torture to get IAX working properly. Yeah, I wasn't referring particularly to the original post, just the way the thread turned against IAX like if it's not a viable solution and my point all along has been that for *us* IAX2 endpoints have worked better and easier to configure than SIP ones. Then it turned into a pissing contest, like you say, it happens in every list with the topic this or that. Again, as I pointed out to Steve above, and after reading all of your responses, our SIP/NAT woes seem obviously ignorance on our part, but that doesn't shadow the fact that IAX2 is working great for us with el-cheapo endpoints like Atcom's AG-188N and I would wish that many more manufacturers supported IAX2. We are happy with IAX and honestly never even had the need/curiosity to deal with the many SIP/NAT problems where sometimes it works great, and other times is a real pain in the ass that takes huge amounts of support to fix, and unhappy customers. On the other hand, IAX took some engineering efforts at first, but the support issues are practically non-existent. -- Alejandro Imass I always posted that my view was based on experience. My nieces and I made a viable home phone system out of strings and paper cups It is a real pain when you grow so large and then have to switch over to SIP, might as well go with an Industry Standard then code that is and has always been broken since it's inception. You will find IAX2 trunking issues dating back to 2005 and all sorts of IAX2 related problems since I started way before Asterisk 1.0. They have never got it right, SIP either, but at least SIP is compliant enough to work just about all the time unless. Try IAX, the predecessor of IAX2. My alternator is currently not charging my battery enough for nightime driving unless I turn off the radio and A/C. It is fine without the extra variables. This is nothing new. Knowing that when the demand rises, my battery will die and the vehicle will falter and eventually stall means I am going to replace the alternator. Say I need my High Beams or to charge something via cig lighter, I will end up stranded and need to take emergency action. I could buy a used alternator, but I have no past experience with it and have no idea how it will perform. My choice of proper course of action is to put in something that is known by all to work, maybe a bad unit, but backed by an immediate exchange. I will replace the battery and inspect other potential problem areas and eliminate them as well. Now I will have averted any problems down the road by doing it the right way rather than hopping along on something that has been borken since day one. If you are going to do the job, do it right from the start so that you can grow or change with ease and use real recognized standards. If you are just playing around, do whatever. Actually do whatever, and learn the hard way, I don't care, just trying to help. Thanks, Steve Totaro -- _ -- 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] Getting Ulimit Message after restart asterisk service
Thanks Danny, I would like to know do I need to worry about this message? And why I'm getting this ulimit message? Please provide reason briefly From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] Getting Ulimit Message after restart asteriskservice To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 00c101ccf6f2$9e8c11c0$dba43540$@debsinc.com Content-Type: text/plain; charset=us-ascii This one is simple. Open /usr/sbin/safe_asterisk and put # in first character of line 86 and 102. Or modfy /etc/sudoers to allow your sudo to execute ulimit. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, February 29, 2012 8:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Ulimit Message after restart asterisk service Hi all, Currently I'm getting this message after restarting asterisk service; Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted Before when I had root access I was not facing this message after that system administrator assigned me sudo access for restarting asterisk service. Please assist me out to resolve this issue at earliest. I also tried to set ulimit till 4-40 times higher than currently set i.e. 1024 but still giving me same message. -- Regards, Ahmed Munir Chohan -- Regards, Ahmed Munir Chohan -- _ -- 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] Getting Ulimit Message after restart asterisk service
The good folks at Asterisk wish to limit the number of open files used by Asterisk to 32K (see line 32). If you aren't a super-user, chances are that Linux will cut you off at a number much less than that anyway. The reason you are getting the message; your user/sudo user can't execute ulimit. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, February 29, 2012 2:47 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Getting Ulimit Message after restart asterisk service Thanks Danny, I would like to know do I need to worry about this message? And why I'm getting this ulimit message? Please provide reason briefly From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] Getting Ulimit Message after restart asteriskservice To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 00c101ccf6f2$9e8c11c0$dba43540$@debsinc.com Content-Type: text/plain; charset=us-ascii This one is simple. Open /usr/sbin/safe_asterisk and put # in first character of line 86 and 102. Or modfy /etc/sudoers to allow your sudo to execute ulimit. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, February 29, 2012 8:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Ulimit Message after restart asterisk service Hi all, Currently I'm getting this message after restarting asterisk service; Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted Before when I had root access I was not facing this message after that system administrator assigned me sudo access for restarting asterisk service. Please assist me out to resolve this issue at earliest. I also tried to set ulimit till 4-40 times higher than currently set i.e. 1024 but still giving me same message. -- Regards, Ahmed Munir Chohan -- Regards, Ahmed Munir Chohan -- _ -- 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 distributions
Are there any particular reasons anybody would cite to choose one over the other? FreePBX have also an ISO distribution - I would recommend to use that one. HTH, Ioan On Wed, Feb 29, 2012 at 7:43 PM, Danny Nicholasda...@debsinc.com wrote: Asterisk Now should serve your needs nicely. -- _ -- 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] asterisk distributions
I would say that this is correct http://support.freepbx.org/forum/freepbx/general-help/freepbx-vs-asterisknow -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett Sent: Wednesday, February 29, 2012 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk distributions Are there any particular reasons anybody would cite to choose one over the other? FreePBX have also an ISO distribution - I would recommend to use that one. HTH, Ioan On Wed, Feb 29, 2012 at 7:43 PM, Danny Nicholasda...@debsinc.com wrote: Asterisk Now should serve your needs nicely. -- _ -- 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] Postgresql in Asterisk
I finally solve the problem, in gentoo the permission of dir /var/run/postgresql/ is: drwxrwx--- 2 postgres postgres 4096 Feb 29 18:09 postgresql so if we want to connect asterisk to postgresql, we need to add the user that runs asterisk to the group postgres and with this finally I can connect with unixODBC to postgresql database I hope this help some one. Regards, On Mon, 2012-02-27 at 13:49 -0600, Sergio Basurto wrote: Thank you Jonathan, I already do the steps you mention, my configuration is: in res_odbc.conf enabled = yes dsn = asterisk-connector pre-connect = yes in odbc.ini [asterisk-connector] Description = PostgreSQL connection to 'asterisk' database Driver = PostgreSQL Database= db_asterisk Servername= localhost UserName= asterisk Password= secret Port= 5432 Protocol= 9.1 ReadOnly= No RowVersioning = No ShowSystemTables= No ShowOidColumn = No FakeOidIndex= No ConnSettings= in odbcinst.ini [PostgreSQL] Description = ODBC for PostgreSQL Driver = /usr/lib/libodbcpsql.so Setup = /usr/lib/libodbcpsql.so FileUsage = 1 if I run with root: #echo select 1 | isql -v asterisk-connector returns +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL select 1 ++ | ?column? | ++ | 1 | ++ SQLRowCount returns 1 1 rows fetched This show me that it can connect, the thing is that in the asterisk logs it returns: res_odbc.c: Connecting asterisk res_odbc.c: res_odbc: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket res_odbc.c: Failed to connect to asterisk res_odbc.c: Registered ODBC class 'asterisk' dsn-[asterisk-connector] res_odbc.c: res_odbc loaded. I notice that if I run the isql command with other user than root, it returns [S1000][unixODBC]Could not connect to the server; Could not connect to remote socket. [ISQL]ERROR: Could not SQLConnect I guess is an extra configuration for ODBC that I am missing, what you think? Regards, On Fri, 2012-02-24 at 13:16 -0600, Jonathan Rose wrote: You need to make sure ODBC is actually getting a connection made with your database. What you should see under ODBC DSN settings: Name: asterisk DSN:asterisk-connector Last connection attempt: WHATEVER Pooled: No/Yes Connected: Yes Connected: Yes is the important part. Remember, you need to have an account in postgres that can be logged into. I made one on my machine with the following: name = asterisk password = secret And in /etc/odbc.ini, I have the following connector established: [asterisk-connector] Description = PostgreSQL connection to 'asterisk' database Driver = PostgreSQL Database= asterisk Servername = localhost UserName= asterisk Password= secret Port= 5432 Protocol= 8.1 I'm guessing this will be 9.1 in your case ReadOnly= No RowVersioning = No ShowSystemTables= No ShowOidColumn = No FakeOidIndex= No ConnSettings= While my res_odbc.conf looks like this: [asterisk] enabled = yes dsn = asterisk pre-connect = yes In addition to having a connector defined, you need to have an ODBC adapter for postgres. I think this might come with ODBC byd efault though. When I was using mysql, I had to get a separate adapter to make it work and set the path to it in Driver. I don't think that is the case with pgsql though. Go ahead and post your extconfig.conf. I'm guessing that the reason you are able to post CDRs in spite of not having the Connected status show up in your ODBC show is because you are connecting with res_pgsql.conf instead of odbc. - Original Message - From: Sergio Basurto sbasu...@soft-gator.com To: asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2012 6:54:47 AM Subject: Re: [asterisk-users] Postgresql in Asterisk On Wed, 2012-02-22 at 06:48 -0600, Sergio Basurto wrote: Hello, I install asterisk an postgresql 9.1 in gentoo, I already did the configuration in both asterisk and postgresql, in fact If I make a call and asterisk log it to CDR table, my question is: I make a typo mistake I mean If I make a call asterisk already log it into CDR table.
[asterisk-users] for help
hello,everyone: i'm a freshman on voip. there is a problem about asterisk . there is a 4E1 with signalling(ss7) and three servers(a part has one server and the other has two server). Two servers on the same part share the same point code as a cluster to get load sharing Then the two different parts can interconnect with each other. I try to edit ss7.conf .but it doesn't work . Only one of the servers on the same part works,the other do nothing. Can sharing one point code with two or more servers? Waiting for your help. ps: software version chan_ss7 2.0.0 asterisk 1.4.26 bai 3.1.2012 -- _ -- 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
