Re: [asterisk-users] Register = plain text password
On Wednesday 22 January 2014, José Pablo Méndez Soto wrote: Hello, Is there anyway to encrypt or scramble a bit the secret used to register with a provider? Im talking about the register = fromuser@fromdomain:secret@host directive in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf No. Well. You *could* scramble it for storage; but that would only lull you into a false sense of security, because ultimately it would have to be able to be unscrambled by a program that was already right there on the machine, somewhere under /usr/src/ where any competent programmer can look at it. The client *has* to know the password in plaintext (or at least, how to decrypt the stored, encrypted password), in order to be able to send it to the server. The way things stand, the configuration file with the password in it need only be readable by the root user. And you know it has a password in it, so you take care with it. Here is an explanation from the developers of the Pidgin IM client, as to why they store passwords in plaintext in their configuration file: https://developer.pidgin.im/wiki/PlainTextPasswords This clever dude modified the code back in 1.4: http://www.oneharding.com/voip/asterisk_md5_register.html Unfortunately, that doesn't work. It just elevates a stolen hash to the same level of usefulness as a stolen password (and she even says so much, in the linked article). I imagine that so many years later, and now with the implementation of pjsip this secret could be better protected? No, because the underlying problem -- that decrypting a stored password also requires the decryption key; but if the decryption key and encrypted password are stored on the same machine, then anyone with access to the machine is able to decrypt the password -- is a limitation of the universe, *not* a limitation of present-day technology. There is simply nothing that anybody could invent that would get around this. It is very unsafe to keep the accounts password right out there. Any ideas? It's hidden behind another password, and that's about as secure as it's mathematically possible ever to make it. And if someone else has root access to your machine, then I humbly suggest that a SIP password might not be the driest lentil you have to soak. -- AJS Answers come *after* questions. -- _ -- 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 11.7.0 not receiving registration from local address
Hi, I face a problem which look like the same as David with his thread Asterisk not receiving call from VPN address. I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM) having IP 192.168.111.14, my phone network is in the range 192.168.10.x I updated lately to 11.7.0 version and now no one of my phones can register anymore to the asterisk. Ngrep as well as wireshark shows the traffic going in on eth0 from VM, but inside Asterisk, nothing, sip set debug ip ip from phones shows nothing. This asterisk is also connected to other trunks we have outside our network, there is no problem here, registration is fine. Problem seems to be only with intranet addresses. We have localnet=192.168.0.0/255.255.0.0, We add permit=0.0.0.0/0.0.0.0, nothing to do :-( Thanks for any hint -- Daniel -- _ -- 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] core show channels truncates channel names?
Thanks that's perfect! I would use the AMI but I have to run against Asterisk 1.4 and the ami command 'coreshowchannels' didn't appear until Asterisk 1.6 From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett [rmudg...@digium.com] Sent: Tuesday, January 21, 2014 6:12 PM To: Asterisk Users List Subject: Re: [asterisk-users] core show channels truncates channel names? On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data) IAX2/FONEMITEL123456 1296197222@entryhomemailto:1296197222@entryhome Ringing AppDial((Outgoing Line)) SIP/200-093e4998s@macro-dialexternalmailto:s@macro-dialexternal Ring Dial(IAX2/FONEMITEL1234567890/ 2 active channels 1 active call How can I get the full username of all active channels? (I realize I can use the AMI but trying to avoid that) Note that the above output is generated on an Ast 1.4 system Use core show channels concise. It was intended for script querying of the channels. However, you really should AMI actions where possible as they are not likely to change from version to version where CLI commands can and do change. Richard -- _ -- 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 11.7.0 not receiving registration from local address
Le 22/01/2014 14:01, Administrator TOOTAI a écrit : Hi, I face a problem which look like the same as David with his thread Asterisk not receiving call from VPN address. I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM) having IP 192.168.111.14, my phone network is in the range 192.168.10.x I updated lately to 11.7.0 version and now no one of my phones can register anymore to the asterisk. Ngrep as well as wireshark shows the traffic going in on eth0 from VM, but inside Asterisk, nothing, sip set debug ip ip from phones shows nothing. This asterisk is also connected to other trunks we have outside our network, there is no problem here, registration is fine. Problem seems to be only with intranet addresses. We have localnet=192.168.0.0/255.255.0.0, We add permit=0.0.0.0/0.0.0.0, nothing to do :-( Thanks for any hint Definitely something wrong with 11.7.0 I switched back to 11.6.0.1 and phone registration is OK. David, you should probably try to downgrade if you have an 11.7.0 installed -- Daniel -- _ -- 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 receiving call from VPN address
Le 20/01/2014 03:51, David Cunningham a écrit : Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. Hi David, if you have a 11.7.0 version try to downgrade to 11.6.0.1 See my post about similar issue with no registration of our intranet phones -- Daniel -- _ -- 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] ISDN Cause Code 47 Errors
We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All the PRIs are setup as individual spans, we are not using NFAS. If the provider sets the hunt method to 'least recently used channel', then I can receive calls on other channels on the secondary span, it is just the first 2 that consistently fail. We have had occasion where the error occurs on first span. If enough calls come in at the same time, callers who happen to land on channel 3 or above are OK. When the problem happens on the first span, if we physically disconnect the first span(RED alarm), the calls hunt to the second span and all calls seem to process properly. The only way to clear the cause 47 errors from the first span is a power cycle on the provider equipment. Power cycling my equipment does not solve the problem, only when I cycle their equipment. The provider says the cause 47 is coming from my equipment, yet the 'core set debug on' log, unless I am reading it wrong, says it is coming from their side. I have a second server as a backup. Both servers have identical hardware and software. When switching to the backup server, the problem remains. I had the same setup on the previous provider, except using NFAS, and did not have this problem. Am I reading the log correctly? Do I have something setup incorrectly? Is there any way to get even lower level debugging on the PRI? Has anyone ever had this problem and if so what was done to resolve it? Thanks in advance for any insight, Dale = general config info = There are two trunk groups for hunting by the provider TG1 = spans 2 and 4, TG2 = spans 1 and 3 I have two dual port cards in the server. Each card has 1 span from each TG so the lost of a card will keep the other span operational. The provider does the same across VWIC cards. The spans do NOT use NFAS. == My equipment/software: == 2x Digium TE210 dual T1/E1 with hw echo canceler and timing cable installed. Asterisk 1.8.12 dahdi 2.5.0.1 libpri 1.4.12 == Provider equipment: == Cisco 2821 with 2 VWIC2-@MFT-T1/E1 === Log sample with ISDN debug on === [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Protocol Discriminator: Q.931 (8) len=87 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent from originator) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Message Type: SETUP (5) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [04 03 80 90 a2] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 User information layer 1: u-Law (34) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [18 03 a9 83 82] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4ChanSel: As indicated in following octets [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4Ext: 1 Coding: 0 Number Specified Channel Type: 3 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4Ext: 1 Channel: 2 Type: CPE] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [1c 1d 9f 8b 01 00 a1 17 02 01 1f 02 01 00 80 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 57 49] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Facility (len=31, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17, 0x02, 0x01, 0x1F, 0x02, 0x01, 0x00, 0x80, 0x0F, 'Cell Phone WI' ] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [28 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 57 49] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Display (len=15) [ Cell Phone WI ] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [6c 0c 21 80 34 31 34 33 33 31 32 34 37 30] [Jan 14 12:56:04]
[asterisk-users] Mailinglist Digium IP-phones : provisioning Digium D70
Hello, is there a mailinglist where I can post questions regarding Digium IP-phones ? I have the following question : I'm trying to provision a Digium D70 IP-phone from a https provisioning server. The Digium D70 contacts the provisioning server correctly but seems to log in with the wrong credentials : /var/log/ssl_access_log : XX.XX.XX.46 - - [22/Jan/2014:12:15:09 +0100] GET /101001/000fd3068c59.cfg HTTP/1.1 401 481 XX.XX.XX.46 - - [22/Jan/2014:12:15:10 +0100] GET /101001/000FD3068C59.cfg HTTP/1.1 401 481 XX.XX.XX.46 - - [22/Jan/2014:12:15:10 +0100] GET /101001/.cfg HTTP/1.1 401 481 XX.XX.XX.46 - - [22/Jan/2014:12:15:11 +0100] GET /101001 HTTP/1.1 401 481 I am absolutely sure that I have given the correct username and password in the Digium D70 phone. I have tried logging in with my Firefox browser to the provisioning server, and that is succesful ! I get asked for the username and password, and I can see the content of the cfg-file. /var/log/ssl_access_log : XX.XX.XX.46 - 101001 [22/Jan/2014:12:32:26 +0100] GET /101001/000fd3068c59.cfg HTTP/1.1 200 2257 The difference I see is the 101001, which is the username. I see the following in the logs of the https provisioning server : [Wed Jan 22 14:00:34 2014] [error] [client XX.XX.XX.46] Digest: client used wrong authentication scheme `Basic': /101001 So how do I get the Digium IP-phone to use the md5 digest authentication ?? 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] ISDN Cause Code 47 Errors
On Mon, Jan 20, 2014 at 5:16 PM, Dale Noll dn...@wi.rr.com wrote: We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All the PRIs are setup as individual spans, we are not using NFAS. If the provider sets the hunt method to 'least recently used channel', then I can receive calls on other channels on the secondary span, it is just the first 2 that consistently fail. We have had occasion where the error occurs on first span. If enough calls come in at the same time, callers who happen to land on channel 3 or above are OK. When the problem happens on the first span, if we physically disconnect the first span(RED alarm), the calls hunt to the second span and all calls seem to process properly. The only way to clear the cause 47 errors from the first span is a power cycle on the provider equipment. Power cycling my equipment does not solve the problem, only when I cycle their equipment. The provider says the cause 47 is coming from my equipment, yet the 'core set debug on' log, unless I am reading it wrong, says it is coming from their side. I have a second server as a backup. Both servers have identical hardware and software. When switching to the backup server, the problem remains. I had the same setup on the previous provider, except using NFAS, and did not have this problem. Am I reading the log correctly? Yes. Asterisk has accepted the selected channel and CONNECTed the call. It is the peer that is disconnecting the call with cause 47. This really appears to be a problem in the providers equipment. Do I have something setup incorrectly? I don't see anything wrong. Is there any way to get even lower level debugging on the PRI? Not really. The problem is at layer 3 (Q.931) not layer 2 (Q.921). Turning on intense PRI debug will add nothing but noise to the debug log. snip === Log sample with ISDN debug on === [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Protocol Discriminator: Q.931 (8) len=87 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent from originator) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Message Type: SETUP (5) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [04 03 80 90 a2] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 User information layer 1: u-Law (34) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [18 03 a9 83 82] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4ChanSel: As indicated in following octets [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4Ext: 1 Coding: 0 Number Specified Channel Type: 3 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4Ext: 1 Channel: 2 Type: CPE] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [1c 1d 9f 8b 01 00 a1 17 02 01 1f 02 01 00 80 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 57 49] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Facility (len=31, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17, 0x02, 0x01, 0x1F, 0x02, 0x01, 0x00, 0x80, 0x0F, 'Cell Phone WI' ] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [28 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 57 49] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Display (len=15) [ Cell Phone WI ] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [6c 0c 21 80 34 31 34 33 33 31 32 34 37 30] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Jan 14 12:56:04] VERBOSE[13262]
Re: [asterisk-users] ISDN Cause Code 47 Errors
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent from originator) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Message Type: RELEASE COMPLETE (90) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 [08 02 80 af] [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Ext: 1 Cause: Resource unavailable, unspecified (47), class = Network Congestion (resource unavailable) (2) ] My guess is your provider did not have a free voice channel to pass audio at some leg in the call. There could be multiple legs in the call and one of them had 'Network Congestion'. [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 Received message for call 0xb750d598 on link 0x9b33f2c TEI/SAPI 0/0 [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 -- Processing IE 8 (cs0, Cause) [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span: 4 q931.c:8567 post_handle_q931_message: Call 6918 enters state 0 (Null). Hold state: Idle [Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] Span 4: Processing event PRI_EVENT_HANGUP [Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] -- Span 4: Channel 0/2 got hangup, cause 47 Richard -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qualify=yes outboundproxy
I am sure I need outboundproxy :) We use Kamailio in front of our softswitch as a type of firewall and because not all sip endpoints can set the headers such that they will work with said softswitch. Calling messages (and calling!) work just fine through the proxy - that setting just doesn't seem to be honored by the qualify=yes messaging. This isn't a particular problem, but when we setup backup trunks that go through a different proxy, for redundancy without the monitoring, it takes a fair amount of time before the calls go through while it waits for the problem trunk to timeout on it's own. -Nick -- _ -- 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] qualify=yes outboundproxy
We use opensips as a type of firewall as well and don't need to set qualify=yes. N -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Asterisk 12
Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm a little lost. I see that Asterisk 12 has a nice REST API - very nice - something I can use. However, and this is gonna sound dumb - but all the CLI commands are different now. What did I miss? Can anyone, please, anyone point me to a good, simple to understand tutorial on the new CLI? I am so, so freaking lost! I'm not looking for hand-holding, I just want to understand. Something that will show me how to: - create users - configure SIP trunks - configure basic dialplan I'm lost - anyone point me to a resource that is easy to follow? Once I get the jist, I think I'll be fine. I looked on http://www.voip-info.org - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? Overwhelming - sigh. Thank much - any help would be appreciated - next time you are in Manchester NH - I'll make you my fave Tequila Sour drink! G -- _ -- 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] Question about Asterisk 12
Maybe it's just me if I'm not mistaken the three things you listed are usually configured using the config files not on CLI. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Wystead Sent: Wednesday, January 22, 2014 3:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about Asterisk 12 Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm a little lost. I see that Asterisk 12 has a nice REST API - very nice - something I can use. However, and this is gonna sound dumb - but all the CLI commands are different now. What did I miss? Can anyone, please, anyone point me to a good, simple to understand tutorial on the new CLI? I am so, so freaking lost! I'm not looking for hand-holding, I just want to understand. Something that will show me how to: * create users * configure SIP trunks * configure basic dialplan I'm lost - anyone point me to a resource that is easy to follow? Once I get the jist, I think I'll be fine. I looked on http://www.voip-info.org http://www.voip-info.org/ - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? Overwhelming - sigh. Thank much - any help would be appreciated - next time you are in Manchester NH - I'll make you my fave Tequila Sour drink! G -- _ -- 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] Question about Asterisk 12
I looked on http://www.voip-info.org - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? https://wiki.asterisk.org/wiki/display/AST/Home To my knowledge the voip-info domain is mostly outdated information these days. The official asterisk wiki is where you want to look for current information. Sorry I can't help you with anything else, I've not had time to play with 12 Yet. -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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 receiving call from VPN address
Hello, We tried Asterisk 1.8 and 1.6, but not yet Asterisk 11. We'll keep it in mind. In the meantime we've decided to try a different network configuration instead, so the VPN network is separated from what Asterisk sees. Thanks for all the advice given. On 23 January 2014 00:42, Administrator TOOTAI ad...@tootai.net wrote: Le 20/01/2014 03:51, David Cunningham a écrit : Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. Hi David, if you have a 11.7.0 version try to downgrade to 11.6.0.1 See my post about similar issue with no registration of our intranet phones -- Daniel -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] type=peer vs type=user (depricated?)
I'm looking at setting type=peer vs type=user (in both IAX and SIP conf entries), and I found a comment attributed to digium (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that type=user is depricated and that we should only use type=peer Is that still correct? Will type=user be phased out, and should even new installs of older asterisk versions (eg: 1.6) use type=peer only? Are people still using type=user for phone sets? (and type=peer for upstream/trunks only) -- _ -- 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] AMI eventmask question
I'm creating an AMI client and I only want to get newchannel events (as well as responses to any actions I initiate). What would I set the eventmask to to only get the newchannel events? For anyone else looking...is there a table somewhere online that maps events to their eventmask categories? I checked the asterisk wiki and voip-info but can't find this... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qualify=yes outboundproxy
We use opensips as a type of firewall as well and don't need to set qualify=yes. As I said, I don't *need* to set qualify=yes for things to work. It's just that trunk failover takes ~30 seconds. If the first/main trunk in an outbound route is down, outbound calls just sit with dead air for a ~30 second timeout before the next trunk in the outbound route is tried. With qualify=yes, if the trunk is down, asterisk will mark it as such and will skip that trunk until it starts responding to the options ping again. -Nick -- _ -- 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] qualify=yes outboundproxy
We use opensips as a type of firewall as well and don't need to set qualify=yes. As I said, I don't *need* to set qualify=yes for things to work. It's just that trunk failover takes ~30 seconds. If the first/main trunk in an outbound route is down, outbound calls just sit with dead air for a ~30 second timeout before the next trunk in the outbound route is tried. With qualify=yes, if the trunk is down, asterisk will mark it as such and not try that trunk until it starts responding to the options ping again. Of course, the fact that it doesn't seem to work properly is the problem. -Nick -- _ -- 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
