Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?
I would be very interested in using sipsak for something like this. What have you tried so far? -Thufir On Mon, 16 Jan 2017, Olivier wrote: Thinking over my previous, I wonder if sipsak could be used to send outgoing SIP NOTIFY messages. Would both Asterisk and sipsak be able to share networks resources ? Thoughts ? 2017-01-16 14:10 GMT+01:00 Olivier <oza.4...@gmail.com>: [..] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial() from the console?
On Wed, 11 Jan 2017, Doug Lytle wrote: On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: Can I dial directly from the asterisk console with the Dial() application? console dial number@context Thanks, that's much more intuitive :) -Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip:p...@noname.com
The SIP trace shows messages from what I took to be a suspicious connection from sip:p...@noname.com so I added that IP address to IP tables...but then anveo showed as unreachable so I removed that rule. Yes, I'm running fail2ban. What are these messages from sip:p...@noname.com? The domain name alone set off alarm bells for me. (I was looking for my own registration attempts when I turned on SIP debugging.) SIP trace: fqdn*CLI> fqdn*CLI> sip set debug on SIP Debugging enabled fqdn*CLI> <--- SIP read from UDP:67.212.84.21:5010 ---> OPTIONS sip:s...@xxx.xxx.xxx.xxx:5060 SIP/2.0 Via: SIP/2.0/UDP 67.212.84.21:5010;branch=0 From: sip:p...@noname.com;tag=uloc-5875e606-bf5-dea1e-52564b36-00fe47a3 To: sip:s...@xxx.xxx.xxx.xxx:5060 Call-ID: cb004ab7-97b14601-e7ade23@67.212.84.21 CSeq: 1 OPTIONS Content-Length: 0 <-> --- (7 headers 0 lines) --- Sending to 67.212.84.21:5010 (NAT) Looking for s in default (domain xxx.xxx.xxx.xxx) <--- Transmitting (NAT) to 67.212.84.21:5010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.212.84.21:5010;branch=0;received=67.212.84.21;rport=5010 From: sip:p...@noname.com;tag=uloc-5875e606-bf5-dea1e-52564b36-00fe47a3 To: sip:s...@xxx.xxx.xxx.xxx:5060;tag=as5f595fce Call-ID: cb004ab7-97b14601-e7ade23@67.212.84.21 CSeq: 1 OPTIONS Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Accept: application/sdp Content-Length: 0 <> Scheduling destruction of SIP dialog 'cb004ab7-97b14601-e7ade23@67.212.84.21' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog 'cb004ab7-90004601-06ade23@67.212.84.21' Method: OPTIONS Reliably Transmitting (NAT) to 67.212.84.21:5010: OPTIONS sip:sip.anveo.com SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK601302be;rport Max-Forwards: 70 From: "asterisk" <sip:aster...@xxx.xxx.xxx.xxx>;tag=as194a0afc To: Contact: <sip:aster...@xxx.xxx.xxx.xxx:5060> Call-ID: 6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4 Date: Wed, 11 Jan 2017 14:56:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:67.212.84.21:5010 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK601302be;rport=5060;received=xxx.xxx.xxx.xxx From: "asterisk" <sip:aster...@xxx.xxx.xxx.xxx>;tag=as194a0afc To: ;tag=a1766e4537c6d6082807422b1789bf43.b9ae Call-ID: 6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060 CSeq: 102 OPTIONS Server: Anv Edge Proxy 3.5 Content-Length: 0 <-> --- (8 headers 0 lines) --- Really destroying SIP dialog '6e15b7534a1b1e852464e02a5fca4...@xxx.xxx.xxx.xxx:5060' Method: OPTIONS fqdn*CLI> sip set debug off SIP Debugging Disabled fqdn*CLI> fqdn*CLI> sip show peers Name/username HostDyn Forcerport ComediaACL Port Status Description anveo/1234567890 67.212.84.21Yes Yes5010 OK (78 ms) demo_alice(Unspecified)D Yes Yes0UNKNOWN demo_bob (Unspecified)D Yes Yes0UNKNOWN piter (Unspecified)D Yes Yes0UNKNOWN thufir(Unspecified)D Yes Yes0UNKNOWN 5 sip peers [Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline] fqdn*CLI> fqdn*CLI> sip show peer anveo * Name : anveo Description : Secret : MD5Secret: Remote Secret: Context : from-anveo Record On feature : automon Record Off feature : automon Subscr.Cont. : Language : Tonezone : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : "" <> MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes Symmetric RTP: Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: No Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost :
[asterisk-users] Dial() from the console?
Can I dial directly from the asterisk console with the Dial() application? or, is channel originate preferred: channel originate SIP/thufir extension 18003569377@outbound thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip show [general]?
I appreciate that the console lets you see the details for a peer with "sip show peer foo". Certainly, I can look in sip.conf to see the [general] context, but can I output those settings, and only those settings, to the console? thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anveo, a different kind of trunk provider?
Anveo has their config as so: [anveo] type=friend host=sip.anveo.com port=5010 username= ACCOUNT_NUMBER secret= SIP_PASSWORD insecure=port,invite disallow=all allow=ulaw context=from-anveo http://www.anveo.com/faq.asp?code=sip_asterisk but this seems slightly odd. I have an account with them where my hard phone, an SPA 942 IP phone, connects directly to them. I just entered the SIP details. Presumably they're running Asterisk and have it configured for my SIP account. But, their registration string with Asterisk is: Locate [general] secion and add the following register => ACCOUNT_NUMBER:sip_passw...@sip.anveo.com:5010 Wouldn't this send every outbound call through that Anveo account? Let's say that I add a more hard or softphones, but configure them to connect to my Asterisk server running on AWS. When Anveo dials out to the POTS everything shows as coming from a single number, the account number? thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rasberry pi
ok, that's really all I need to know. Of course, if anyone else wants to throw in their two cents, don't let me stop you :) -Thufir On Wed, Jul 6, 2016 at 1:36 AM, Frank Vanoni <mailingl...@linuxista.com> wrote: > I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with > Ubuntu Server 14.04. > > Works fine! :-) > > Frank > > On Wed, 2016-07-06 at 01:10 -0700, Thufir wrote: > > I'm debating between a cloud PBX or, perhaps, rasberry pi. For a > > SOHO, maybe three hardphones, rasberry pi would suffice? I would be > > amazed, but, if so, great. > > > > > -- > _ > -- 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] what is a SIP invite, and who can issue them?
On Wed, 29 Jun 2016 06:38:53 -0300, Joshua Colp wrote: > An INVITE is a request to set up a session, commonly referred to as a > call. Anything supporting SIP to establish calls uses INVITE to do so. > It's equivalent to picking up the phone and dialing a number. an INVITE would never be sent unless a call, or other communication with an endpoint, was being attempted? thanks, Thufir -- _ -- 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] rasberry pi
I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO, maybe three hardphones, rasberry pi would suffice? I would be amazed, but, if so, great. thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to read sip debug
This is interesting: "Note that the To and From header fields are not reversed in the response message as one might expect them to be. This is because the To and From header fields in SIP are defined to indicate the direction of the request, not the direction of the message. " -Cisco so, when I'm receiving an inbound call, the direction would be telnyx first, then me. Regardless of whether the ?message? is from me or the provider. -Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to read sip debug
Generally, what am I looking for when turning SIP debug on? More specifically, the provider says that I'm returning a 404 when they try to call me. Now, I had inbound working, literally, the other day. Outbound works fine. I "may" have broken it either through Asterisk config or the providers portal with settings. Ok, I broke it -- not sure how. comments interspersed: mordor*CLI> Reliably Transmitting (NAT) to 192.76.120.10:5060: I think/infer/assume that this is the IP address for telnyx SIP servers OPTIONS sip:sip.telnyx.com SIP/2.0 What does OPTIONS mean? Via: SIP/2.0/UDP :5060;branch=z9hG4bK28142189;rport rport relates to NAT? The message is via SIP UPD from my externip what is branch? Max-Forwards: 70 70 hops max? From: "asterisk" <sip:asterisk@>;tag=as1a7aca46 from my externip, with a hash to keep the calls straight? To: easy, to telnyx Contact: <sip:asterisk@:5060> from me Call-ID: 6fce72627f253b7f2e15dac713b52392@:5060 another hashcode, Call-ID ? CSeq: 102 OPTIONS ? User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3 easy enough, my system Date: Wed, 06 Jul 2016 02:17:12 GMT easy, date Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE enumerating accepted replies? Supported: replaces ? Content-Length: 0 no data, just "hi" --- mordor*CLI> If I see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions in a SIP trace, that's relatively clear. But what am I looking for with regards to receiving calls? thanks, Thufir -- _ -- 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] what is a SIP invite, and who can issue them?
I don't understand what a SIP invite is. Certainly it's explained as:
"This article explains the main fields included in a SIP INVITE, which is
sent to set-up a VoIP call. A SIP INVITE message contains typically between
4 and 6 header entries with contact information inside them."
http://www.3cx.com/blog/voip-howto/sip-invite-header-fields/
The article enumerates the headers and explains them. But what sends the
invite? Asterisk? A soft-phone?
I found sample config's to send an invite with Asterisk but no other method
was given. Can only Asterisk send an invite? Why? The article says that
it's sent "to set-up a VoIP call," so presumably any reasonable soft-phone
sends these invites as a normal process.
That's all well and good, but how do send an actual invite and get a
response? This can only be done through Asterisk?
This is in the context of:
Requires IP Authentication to be setup through the portal and associated
with LRN under Telephone Data
<https://portal.telnyx.com/#/app/telephone-data> Tab
Send a SIP Invite to *lrnlookup.telnyx.com <http://lrnlookup.telnyx.com>*
with the number you wish to dip on port 5060
The response will be a SIP 302 redirect for example:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP
172.16.1.12;branch=z9hG4bKfae8cb69f547b8cb;received=172.16.0.179
From: <sip:55@172.16.1.12>;tag=102
To: <sip:55@174.36.199.131>
call-id: 0704037283648236478326200101@172.16.1.12
CSeq: 1 INVITE
Contact: Transfer <sip:55;*rn=+156;npdi;*@174.36.199.131>
Content-Length: 0
If a number has been ported the response will contain the dip indicator
("npdi;") as well as the LRN (rn=+1..), otherwise these fields will be
missing
from https://apidocs.telnyx.com/
and then clicking "Data API" and then "SIP request" for details.
I have a running instance of Asterisk. I would have to handle the invite
through Asterisk and keep it running to make and receive calls? Presumably
this invite is interacting with Asterisk, or something similar, at
telnyx.com -- which seems overkill.
thanks,
Thufir
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Re: [asterisk-users] dial out with channel variable; sub-string usage
On 15-04-09 12:06 PM, Chad Wallace wrote:
but don't know where to put those lines. I have BABY defined as
channel variable:
BABY = SIP/babytel_out
but that seems circular, somehow.
You put them in the context for your clients... From what you show
below, I'd say they go in the local_200 context. You can verify
this by looking in sip.conf, in the section that starts with [200],
find the line that starts with context=. It's probably
context=local_200. Then you put the outbound dialplan in that context
in extensions.conf. Mind you, then 200 is the only phone that can dial
out. 201 can only dial 200 and nothing else.
Wait a minute, slow down. I re-installed, same sort of problem:
vici:~ #
vici:~ # asterisk -rx sip show peers
Name/username Host Dyn Forcerport ACL Port Status
300/300 (Unspecified) D N 0UNKNOWN
301/301 192.168.0.24 D N 5060 OK (29
ms)
302/302 (Unspecified) D N 0UNKNOWN
gs102/gs102 (Unspecified) D N 0UNKNOWN
testcarrier/19876543210 198.38.7.34
N 5065 OK (82 ms)
5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0
offline]
vici:~ #
vici:~ # asterisk -rx sip show peer testcarrier
* Name : testcarrier
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context : default
I simply want all outbound calls to go through a specific context, I
think, if I understand how the channel passes control to the correct
context. I want, or need, an outbound context?
I know that the text has an example of ServerA routing through serverB.
Simply add a line, like:
exten = _9x.,1,Dial(SIP/${EXTEN:1}@babytel_out)
or
exten = _9x.,1,Dial(${BABY}/${EXTEN:1})
Which just brings me back to...the context for BABY...is...? BABY is a
channel variable. In the above CLI output, the channel variable is
testcarrier, which has a context of default.
Each channel variable maps to at most one context? Many channel
variables can map to a single context?
thanks,
Thufir
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[asterisk-users] dial out with channel variable; sub-string usage
I want to do something like:
exten = _NXXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _Nxx,1,Dial(${BABY}/${EXTEN})
exten = _1NXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _011.,1,Dial(Dial({TOLL}/${EXTEN})
exten = _9NXXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _9Nxx,1,Dial(${BABY}/${EXTEN})
exten = _91NXXNxx,1,Dial(${BABY}/${EXTEN})
exten = _9011.,1,Dial(Dial({TOLL}/${EXTEN})
(adapted from the book)
but don't know where to put those lines. I have BABY defined as channel
variable:
BABY = SIP/babytel_out
but that seems circular, somehow.
inbound calls work fine:
[inbound-calls]
exten = 16046289850,1,Dial(SIP/200)
[local_200]
exten = _9x.,1,Set(CALLERID(all)=Ali Baba 123456789)
exten = _9x.,1,Dial(SIP/${EXTEN:1}@babytel_out)
exten = 201,1,Dial(SIP/201)
[local_201]
exten = 200,1,Dial(SIP/200)
in local_200, that just seems suspect. Yes, dial out, but shouldn't it
be using BABY? I don't understand why it's using sub-string with the 1.
thanks,
Thufir
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[asterisk-users] exten versus EXTEN
p 176 has exten = 1NXXNXXX,1,Dial(SIP/${EXTEN}@myprovider)
how is exten distinct from EXTEN? What is this line of code doing?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
says that EXTEN is the current extension.
In ruby, you this:
H = Hash[a = 100, b = 200]
The = is a mapping, or at least that's my understanding. What does it
mean in Asterisk? I didn't
fully appreciate that Asterisk is, apparently, its own language.
thanks,
Thufir
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Re: [asterisk-users] [OT] switches
On Mon, 23 Mar 2015 10:11:54 +, Lukasz Sokol wrote: No, ethernet switch works at lower / physical / MAC layer, NAT is 'above' that; so as long as everything is OK with your TCP/IP settings everywhere, a switch is entirely transparent to TCP/IP (or generally, when it's encapsulated into MAC traffic). so how does a client pc find the server if there's no NAT? by IP address?? That makes no sense, to me, if the switch isn't assigning addresses. -Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trying to connect to asterisk with softphone (logs, etc)
,
ACCOUNT_ICON_PATH=resources/images/protocol/irc/irc32x32.png,
AUTO_CHANGE_USER_NAME=true, CHAT_ROOM_PRESENCE_TASK=true,
NO_PASSWORD_REQUIRED=false, ACCOUNT_UID=IRC:201@192.168.0.99:6697,
SERVER_ADDRESS=192.168.0.99, USER_ID=201, DEFAULT_ENCRYPTION=true,
PROTOCOL_NAME=IRC, ENCRYPTED_PASSWORD=/hcTkghmfRJWFXrWaKDMmA==,
CONTACT_PRESENCE_TASK=true}
java.lang.IllegalArgumentException: nick name contains invalid
characters: only letters, digits and -, \, [, ], `, ^, {, }, |, _ are
allowed
at
net.java.sip.communicator.impl.protocol.irc.IdentityManager.checkNick(IdentityManager.java:194)
at
net.java.sip.communicator.impl.protocol.irc.IrcStack$ServerParameters.init(IrcStack.java:354)
at
net.java.sip.communicator.impl.protocol.irc.IrcStack$ServerParameters.init(IrcStack.java:311)
at
net.java.sip.communicator.impl.protocol.irc.IrcStack.init(IrcStack.java:89)
at
net.java.sip.communicator.impl.protocol.irc.ProtocolProviderServiceIrcImpl.initialize(ProtocolProviderServiceIrcImpl.java:149)
at
net.java.sip.communicator.impl.protocol.irc.ProtocolProviderFactoryIrcImpl.createService(ProtocolProviderFactoryIrcImpl.java:136)
at
net.java.sip.communicator.service.protocol.ProtocolProviderFactory.loadAccount(ProtocolProviderFactory.java:983)
at
net.java.sip.communicator.service.protocol.AccountManager.doLoadStoredAccounts(AccountManager.java:204)
at
net.java.sip.communicator.service.protocol.AccountManager.loadStoredAccounts(AccountManager.java:446)
at
net.java.sip.communicator.service.protocol.AccountManager.runInLoadStoredAccountsThread(AccountManager.java:562)
at
net.java.sip.communicator.service.protocol.AccountManager.access$100(AccountManager.java:26)
at
net.java.sip.communicator.service.protocol.AccountManager$2.run(AccountManager.java:487)
The nick name is 201, no special characters...
I'm on an older, so want to use Jitsi because it's cross platform.
thanks,
Thufir
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Re: [asterisk-users] [OT] switches
On Fri, 13 Mar 2015 20:33:13 -0500, Brian Franklin wrote: If your phones support PoE, I have had huge success with Zyxel: http://www.amazon.com/ZyXEL-ES1100-16P-16-Port-Ethernet-Unmanaged/dp/B00 5GRETMM/ref=sr_1_3?ie=UTF8qid=1426296572sr=8-3keywords=zyxel+poe If you want to go even cheaper, I have successfully used these as well: http://www.amazon.com/TRENDnet-8-Port-100Mbps-Switch-TPE-S44/dp/B000QYEN 1W/ref=sr_1_10?ie=UTF8qid=1426296706sr=8-10keywords=poe+8-port Brian Franklin NTG, Inc. - Problem Solved This is the router/modem gateway the ISP supplied: http://www.cisco.com/web/consumer/support/modem_DPC3825.html When I connect one of these switches to the router, that doesn't create a double-NAT problem? thanks, Thufir -- _ -- 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] UNREACHABLE peer
On 15-03-20 6:42 AM, thufir wrote: I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. Unreachable generally means you have qualify=yes, but the peer is ignoring OPTIONS requests. http://forums.asterisk.org/viewtopic.php?f=13t=92485 but, how do I **know** that, or establish it as fact? thanks, Thufir -- _ -- 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] UNREACHABLE peer
On 15-03-20 6:55 AM, dotnetdub wrote: Turn on sip debugging for this peer and watch for the options sending and response. If you are getting a response to your options asterisk shouldn't be marking the peer as unavailable. is your asterisk behind a firewall? this fits with tech support saying that I was sending options, if that means INVITE et. al. as below: linux-k7qk*CLI [Mar 20 10:31:01] Reliably Transmitting (NAT) to 198.38.7.11:5060: OPTIONS sip:sip.babytel.ca SIP/2.0 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK5311a3f3;rport Max-Forwards: 70 ... CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.29.0-vici Date: Fri, 20 Mar 2015 14:31:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Mar 20 10:31:02] == Manager 'sendcron' logged on from 127.0.0.1 [Mar 20 10:31:02] == Manager 'sendcron' logged off from 127.0.0.1 [Mar 20 10:31:02] Retransmitting #1 (NAT) to 198.38.7.11:5060: OPTIONS sip:sip.babytel.ca SIP/2.0 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK5311a3f3;rport Max-Forwards: 70 ... CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.29.0-vici Date: Fri, 20 Mar 2015 14:31:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Mar 20 10:31:03] Retransmitting #2 (NAT) to 198.38.7.11:5060: OPTIONS sip:sip.babytel.ca SIP/2.0 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK5311a3f3;rport Max-Forwards: 70 ... To: sip:sip.babytel.ca .. CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.29.0-vici Date: Fri, 20 Mar 2015 14:31:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Mar 20 10:31:04] Retransmitting #3 (NAT) to 198.38.7.11:5060: OPTIONS sip:sip.babytel.ca SIP/2.0 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK5311a3f3;rport Max-Forwards: 70 ... CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.29.0-vici Date: Fri, 20 Mar 2015 14:31:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- linux-k7qk*CLI exit linux-k7qk:~ # linux-k7qk:~ # thanks, Thufir -- _ -- 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] UNREACHABLE peer
On 15-03-20 6:55 AM, dotnetdub wrote: Turn on sip debugging for this peer and watch for the options sending and response. If you are getting a response to your options asterisk shouldn't be marking the peer as unavailable. is your asterisk behind a firewall? I have a Cisco DPC3825 DOCSIS 3.0 Data Gateway ... with firewall at low for SPI. I don't recall the other setting, but I've tested it by making VoIP calls with Jitsi configured with this exact peer. Of course, the config isn't exactly the same, but it's the same account, same company, etc. turn on debug like so: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information I'll try that, thanks :) -Thufir -- _ -- 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] UNREACHABLE peer
I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI linux-k7qk*CLI sip show peer testcarrier * Name : testcarrier Secret : Set MD5Secret: Not set Remote Secret: Not set Context : from-trunk Subscr.Cont. : Not set Language : en AMA flags: Unknown Netborder CPD: No Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 1234567890 FromDomain : sip.babytel.ca Port 5060 Callgroup: Pickupgroup : MOH Suggest : default Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: Yes TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No Outb. proxy : nat5.babytel.ca DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : sip.babytel.ca Addr-IP : 198.38.7.11:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 1234567890 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No Status : UNREACHABLE Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No linux-k7qk*CLI thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP show peers: UNREACHABLE
Page 176 of Asterisk, the definitive manual, discusses Connecting an Asterisk system to a SIP provider in the context of, at least the concept of, trunking. Previously, I wasn't able to connect to the peer, and attributed that to a combination of double NAT (plus), and latency and lag due to wi-fi. Now that I'm directly connected to the cable modem (well, gateway router and modem combo), the connection is better and I'm able to make outgoing VoIP calls with Jitsi. Am I right in thinking that the very same connection parameters I entered in Jitsi will work fine when entered in Asterisk with syntax like: register = username:passw...@your.provider.tld and by creating the peer entry in sip.conf for the service provider. One concern is that the provider uses: 1. User ID can be any one of your 11-digit babyTEL telephone numbers. Typically your main number but can be any one of your other phone numbers. 2. For your protection the SIP Password field does not reveal your password until you explicitly click on ‘Show password’. 3. If Outbound Proxy is not supported on your system, try one of the following two options: 1. Add the line “198.38.7.34 sip.babytel.ca” to your system’s “hosts” file and configure the SIP Proxy as: “sip.babytel.ca:5065”. This uses the TCP/IP “hosts” file address mapping mechanism to redirect SIP traffic to the Outbound Proxy. 2. Configure the SIP Proxy as: “198.38.7.34:5065”. This replaces the SIP Proxy address with a resolved Outbound Proxy address. On a mac, I added that line to the hosts file -- but I'm not sure it's required. How do I specify the SIP proxy and the outbound proxy? What's the distinction between a SIP proxy and outbound proxy? In Jitsi, I configured as 123456...@sip.babytel.ca for SIP id. In Connection I used sip.babytel.ca for the registrar and the user, 1234567890, as the the authorization name. I put the proxy as nat5.babytel.ca, port 5065 and the preferred transport as UDP. I don't see all those options, particularly surrounding the proxy and outbound proxy. Again, I'm unclear on why there's a proxy specificed, and then a different outbound proxy is specified as well. How do I establish that I've entered the parameters correctly in Asterisk? Or, how do I establish that the parameters are incorrectly entered? Because Jitsi is able to call out and in, I believe that eliminates NAT, firewall or other networking issues. thanks, Thufir -- _ -- 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] switching from SIP to Skype..or not
On Fri, 13 Mar 2015 11:46:21 -0400, Ron Wheeler wrote: The problem has always been great sound from the other telephone and choppy sound (dropped sound fragments) from me to the caller with only one call going through Asterisk and the network pretty much dedicated to the my workstation. This has survived upgrades of everything (firewall, Asterisk server, workstation) well, that was my thinking -- hardware. If you have just a SIP *client*, ekiga, what-have-you, can it connect out with SIP to SIP fine? because, if so, that would be a powerful litmus test. If that test works, that establishes it's not the network. (Yes, I know you tested bandwidth already, but I'd at least try SIP to SIP client to see if it matches Skype.) Once you know that SIP to SIP works, speaking for myself, I'd just do a clean install. If you've run out of troubleshooting steps, that's the one to use. HTH, Thufir -- _ -- 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] switching from SIP to Skype..or not
On Thu, 12 Mar 2015 10:04:08 -0400, Andres wrote: On 3/12/15 9:39 AM, Ron Wheeler wrote: Your characterization may be true but Skype works much better than SIP when it comes to sound quality. SIP is not to blame for this. Its the audio codec being used. Skype has spend a great deal of effort with their SILK codec by making it highly tolerant of packet loss and jitter. The same cannot be said for the standard codecs Asterisk uses. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. The thing to remember about Skype is that they started out as the small guy, and they had some very interesting ideas, IMHO. I don't actually know it's a sound quality issue, per say. It's double+ NAT, with a wi-fi bridge, plus, sometimes, another wi-fi network. In that situation, skype works from a cell phone! Granted, there are dropped calls, but, eh. The way things stand, I can't, unfortunately, use Ekiga to connect to the **outside** SIP provider because, apparently, there are too many hops: http://superuser.com/questions/880705/ IAX might be useful in this circumstance :) -Thufir -- _ -- 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] switching from SIP to Skype..or not
On Thu, 12 Mar 2015 12:52:46 +, Thufir wrote: Heh, well, I guess it's dead: http://www.digium.com/en/products/software/skype-for-asterisk is this current? http://www.remsys.com/blog/skype-connect-to-asterisk it doesn't solve, I think, the problem I have that SIP clients, sans Asterisk, cannot connect out due to too many hops/bad connection. Only Skype is able, from home at least, to connect out. From what I can tell. -Thufir -- _ -- 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] switching from SIP to Skype..or not
I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok. Is swapping out SIP for Skype a big deal? Heh, well, I guess it's dead: http://www.digium.com/en/products/software/skype-for-asterisk If I have a really bad connection, can I downgrade SIP somehow? I don't really need to use to make voice calls. Or, more specifically, quality, echo, distortion aren't relevant. Just SIP to SIP hello. When I connect to any SIP provider, ekiga, etc, without using Asterisk, I get too many hops errors. While I have another computer on the LAN I can connect to, it's not quite the same. Any thoughts? thanks, Thufir -- _ -- 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] func_odbc 123
with func_odbc, in the definitive asterisk guide, they were suggesting the possibility that part, or perhaps all of, the dialplan could be written as SQL statement!? First off, that sounds like a good idea to me, but the tone of the authors was suggesting not so much, but that it was a personal preference. From a naive perspective, why SQL statements at all? Why not just database config and data instead? thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] switches
On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote: For a very basic setup it would work, but I would suggest POE at a minimum, and vlan support if possible. Gigabit uplinks, 10/100 for the poe ports http://www.amazon.com/NETGEAR-ProSAFE-M4100-D10-POE-Ethernet-Managed/dp/ B00AUEYX0Y/ref=sr_1_3?ie=UTF8qid=1424462577sr=8-3keywords=netgear+poe and Gigabit all ports Hypothetical: lag, choppy connection, dropped calls. Of course, I'd start with checking logs. How would I establish that the problem is that (some) of the ports aren't gigabit? Small office, about five agents. thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] switches
On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote: For a very basic setup it would work, but I would suggest POE at a minimum, and vlan support if possible. thanks for the recomendations :) -Thufir -- _ -- 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] dialplan contexts syntax and terminology
I'm looking into the dialplan specifics:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=DAHDI/r1; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/111222:passw...@iax.binfone.com ; IAX trunk
interface
SIPtrunk=SIP/1234:passw...@sip.provider.net ; SIP trunk
#include extensions-vicidial.conf
Firstly, what language or format is this? Bash script?
the line #include ... what is this called? An include statement?
The [globals] -- what's the terminology for this? It's a context? And
a context is a logical separation in the dialplan? Is that, in any way,
analogous to a function or method?
Once you create your this logical separation, what's the syntax
surrounding invoking a specific context? For example:
tleilax:~ #
tleilax:~ # tail /etc/asterisk/extensions-vicidial.conf
[vicidial-auto]
exten = h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-
NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-
${ANSWEREDTIME})
include = vicidial-auto-internal
include = vicidial-auto-phones
include = vicidial-auto-external
; END OF FILELast Forced System Reload: 2015-02-20 16:49:28
tleilax:~ #
when the above contexts are included, these contexts are declared within
the extensions-vicidial.conf, meaning that when they're declared, they're
not actually used/invoked/called **until** the actual include = foo
syntax?
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Re: [asterisk-users] dialplan contexts syntax and terminology
On Sun, 22 Feb 2015 08:32:26 +0530, Mitul Limbani wrote: READ READ READ I know, I have the 4th edition and I've been reading it. Personally, I find it more general than specific, but I'll go back through that chapter, absolutely. thanks, Thufir -- _ -- 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] 101 called 102 success :)
I called 102 from 101 successfully! I have everything connected to my home router. Asterisk is running on tleilax, so I used my Android phone to call doge. Worked like a charm. I'd been thinking that the firewall was blocking connections, but not at all. Anyhow, thanks to everyone who's help me out. I'm sure I'll have other problems, but huge milestone. -Thufir -- _ -- 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] sipsak 200 for a user, but 404 for a different user...why?
What's the difference between user 123 and devries? Based on the output here, they seem the same..? tleilax*CLI tleilax*CLI sip show users Username Secret Accountcode Def.Context ACL Forcerport 201password 201 default No Yes 123password 123 default No Yes devriespassword devries default No Yes babytelhjkgk58757 default No Yes gs102 X58sKpZCcDfcGT0 gs102 default No Yes tleilax*CLI tleilax*CLI sip show user 123 * Name : 123 Secret : Set MD5Secret: Not set Context : default Language : en Accountcode : 123 AMA flags: Unknown Netborder CPD: No Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : 123 123 ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No tleilax*CLI tleilax*CLI sip show user devries * Name : devries Secret : Set MD5Secret: Not set Context : default Language : en Accountcode : devries AMA flags: Unknown Netborder CPD: No Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : devries 999 ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No tleilax*CLI tleilax*CLI exit tleilax:~ # tleilax:~ # exit logout Connection to tleilax closed. thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:123@tleilax [sudo] password for thufir: No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax message received: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.1.1:55238;branch=z9hG4bK.3e59b63f;alias;received=192.168.1.3;rport=55238 From: sip:sipsak@127.0.1.1:55238;tag=1e6fe4eb To: sip:123@tleilax;tag=as7dc4727d Call-ID: 510649579@127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.32.1-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:192.168.1.2:5060 Accept: application/sdp Content-Length: 0 ** reply received after 0.627 ms ** SIP/2.0 200 OK final received thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:devries@tleilax No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax message received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.1.1:54969;branch=z9hG4bK.38ee5c41;alias;received=192.168.1.3;rport=54969 From: sip:sipsak@127.0.1.1:54969;tag=6e148be1 To: sip:devries@tleilax;tag=as2b617a9b Call-ID: 1846840289@127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.32.1-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 ** reply received after 0.648 ms ** SIP/2.0 404 Not Found final received thufir@doge:~$ thanks, Thufir -- _ -- 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] LAN sip-to-sip
On Mon, 16 Feb 2015 16:12:04 -0500, John Novack wrote: My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet wirenut You SHOULD be able to communicate between devices on the LAN without any firewall issue. I think I might be doing this in a very stupid way. I'm reading Asterisk the definitive guide, but it's very general. Can you describe your, or a typical setup, in a bit more detail? The setup I will use in these notes is this: Asterisk is installed on the gateway/router to the Internet and Ekiga is installed on an 'inside' workstation. http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client What I have is everything connected into the gateway: 192.168.1.1 router 192.168.1.2 tleilax asterisk server; static ip 192.168.1.x doge, client pc; usually .3 Tleilax needs at least two NIC's? One to connect to the gateway, and then perhaps doge directly connects to tleilax, or, there's a switch between doge and tleilax so that other clients can also connect to tleilax. I can't find much in the Asterisk book on this. On all sorts of complex network setups, yes, but not something basic like this. thanks, Thufir -- _ -- 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] connecting with Ekiga; diagnostic tools
I think I'm able to connect with Ekiga, at least it reports registered. Curiously, when I exit Ekiga and switch to SFLphone, it isn't able to connect with the exact same parameters; it just says trying and never resolves. I'm not able to test outside connectivity because of too many hops: thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thu...@ekiga.net -m hi No SRV record: _sip._tcp.ekiga.net No SRV record: _sip._udp.ekiga.net using A record: ekiga.net Max-Forwards set to 0 message received: SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 192.168.1.3:44370;branch=z9hG4bK.6f1c2f33;rport=44370;alias;received=96.48.128.162 From: sip:sipsak@127.0.1.1:44370;tag=981cae4 To: sip:thu...@ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.bf6c Call-ID: 159501028@127.0.1.1 CSeq: 1 OPTIONS Server: Kamailio (1.5.3-notls (i386/linux)) Content-Length: 0 ** reply received after 161.445 ms ** SIP/2.0 483 Too Many Hops final received thufir@doge:~$ but that's ok. How can I test, I mean make a voice call, given that I only have two computers to work with at the moment? The server runs Asterisk on tleilax, and doge is the client. Both connect to the same router. the ip address for tleilax is 192.168.1.2 and the ip address for doge is 192.168.1.3 (generally; doge uses DHCP). When I get more pc's, I can maybe have doge call another pc on the network, but, for right now, what can I do to test this out? thanks, Thufir -- _ -- 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] sipsak 200 for a user, but 404 for a different user...why?
On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: A sip set debug on will give you more info on why you are getting the 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI tleilax*CLI sip set debug on SIP Debugging enabled tleilax*CLI on doge: thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:devries@tleilax -m hi No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax Max-Forwards set to 0 message received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.1.1:56377;branch=z9hG4bK.0edaada3;alias;received=192.168.1.3;rport=56377 From: sip:sipsak@127.0.1.1:56377;tag=6b540010 To: sip:devries@tleilax;tag=as02b0fdd6 Call-ID: 1800667152@127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.32.1-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 ** reply received after 0.844 ms ** SIP/2.0 404 Not Found final received thufir@doge:~$ However, I'm sure you're right that it's the dialplan; I'm looking into it. thanks, Thufir -- _ -- 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] sipsak 200 for a user, but 404 for a different user...why?
On Fri, 20 Feb 2015 17:11:53 -0500, Andres wrote: I don't think so. But you should also see the SIP messages on the console (sip set debug on) without having to look at the log file. Maybe something in your logger.conf is messed up. that worked :) tleilax*CLI tleilax*CLI [Feb 20 21:06:19] --- SIP read from UDP:192.168.1.3:44226 --- OPTIONS sip:345@tleilax SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:44226;branch=z9hG4bK.508a6d72;rport;alias From: sip:sipsak@127.0.1.1:44226;tag=2a099edc To: sip:345@tleilax Call-ID: 705273564@127.0.1.1 CSeq: 1 OPTIONS Contact: sip:sipsak@127.0.1.1:44226 Content-Length: 0 Max-Forwards: 0 User-Agent: sipsak 0.9.6 Accept: text/plain - [Feb 20 21:06:19] --- (11 headers 0 lines) --- [Feb 20 21:06:19] Looking for 345 in trunkinbound (domain tleilax) [Feb 20 21:06:19] --- Transmitting (NAT) to 192.168.1.3:44226 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.1.1:44226;branch=z9hG4bK.508a6d72;alias;received=192.168.1.3;rport=44226 From: sip:sipsak@127.0.1.1:44226;tag=2a099edc To: sip:345@tleilax;tag=as5d21da5c Call-ID: 705273564@127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.29.0-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:192.168.1.2:5060 Accept: application/sdp Content-Length: 0 [Feb 20 21:06:19] Scheduling destruction of SIP dialog '705273564@127.0.1.1' in 32000 ms (Method: OPTIONS) [Feb 20 21:06:38] Really destroying SIP dialog '1876256264@127.0.1.1' Method: OPTIONS [Feb 20 21:06:51] Really destroying SIP dialog '705273564@127.0.1.1' Method: OPTIONS tleilax*CLI exit tleilax:~ # I would've liked to see the hi message, but it's good to see that result server side. -Thufir -- _ -- 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] sipsak 200 for a user, but 404 for a different user...why?
On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: This is showing nothing so I don't think your test message even made it here. I think it looped in the 'doge' server. I was wondering the same thing :) in tleilax, I looked in /var/log/asterisk/messages and see: [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] --- SIP read from UDP:192.168.1.3:38154 --- OPTIONS sip:345@tleilax SIP/2.0 Via: SIP/2.0/UDP 127.0.1.1:38154;branch=z9hG4bK.77fd156e;rport;alias From: sip:sipsak@127.0.1.1:38154;tag=4653e713 To: sip:345@tleilax Call-ID: 1179903763@127.0.1.1 CSeq: 1 OPTIONS Contact: sip:sipsak@127.0.1.1:38154 Content-Length: 0 Max-Forwards: 0 User-Agent: sipsak 0.9.6 Accept: text/plain it seems to work, in that I get 200 OK, with success reflected, apparently, in the log, provided that its numerical. I just changed it from piter to 345 and get success (well, at this at least). This probably has something to do with my dialplan.. Is the message, hi, logged anywhere? -Thufir -- _ -- 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] ports, routers and firewalls
I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not even a call. Ring? Beep? Ping? Some sort of hello world connection. 192.168.1.1 netgear router 192.168.1.2 asterisk (vicidial) 192.168.1.3 ubuntu client 192.168.1.4 mac OSX client (not shown) Do I have a firewall problem which would impact a soft phone from establishing a connection? thufir@doge:~$ thufir@doge:~$ thufir@doge:~$ nmap 192.168.1.1 Starting Nmap 6.46 ( http://nmap.org ) at 2015-02-18 06:10 PST Nmap scan report for 192.168.1.1 Host is up (0.0086s latency). Not shown: 994 closed ports PORT STATE SERVICE 23/tcpopen telnet 53/tcpopen domain 80/tcpopen http /tcp open dec-notes /tcp open freeciv 49152/tcp open unknown Nmap done: 1 IP address (1 host up) scanned in 0.14 seconds thufir@doge:~$ thufir@doge:~$ nmap 192.168.1.2 Starting Nmap 6.46 ( http://nmap.org ) at 2015-02-18 06:10 PST Nmap scan report for 192.168.1.2 Host is up (0.00027s latency). Not shown: 997 filtered ports PORTSTATE SERVICE 22/tcp open ssh 80/tcp open http 443/tcp open https Nmap done: 1 IP address (1 host up) scanned in 4.95 seconds thufir@doge:~$ thufir@doge:~$ thufir@doge:~$ ssh thufir@192.168.1.2 Password: Last login: Mon Feb 16 00:43:01 2015 from 192.168.1.2 Thank you for installing ViciBox Server v.6.0! This software is available for free download at http://www.vicibox.com. If you paid for this software you have been ripped off. Please report any fraud or abuses of this software to ab...@vicidial.com. Please report any bugs on the forum at http://www.vicidial.org To configure the LAN settings type: yast lan To change the server IP in the database type: /usr/share/astguiclient/ADMIN_update_server_ip.pl Official paid-for ViciDial support is available at http://www.vicidial.com Free community-based ViciDial Support is available at http://www.vicidial.org/VICIDIALforum - ViciBox Redux v.6.0.3-141118 Could not chdir to home directory /home/thufir: No such file or directory thufir@tleilax:/ thufir@tleilax:/ nmap 192.168.1.3 Starting Nmap 6.40 ( http://nmap.org ) at 2015-02-18 09:14 EST Nmap scan report for 192.168.1.3 Host is up (0.00075s latency). Not shown: 998 closed ports PORT STATE SERVICE 22/tcp open ssh 2000/tcp open cisco-sccp Nmap done: 1 IP address (1 host up) scanned in 0.15 seconds thufir@tleilax:/ thufir@tleilax:/ thanks, Thufir -- _ -- 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] Respond with 200 OK on OPTIONS
On Tue, 17 Feb 2015 08:28:31 -0600, Matthew Jordan wrote: Asterisk attempts to look up who the OPTIONS request is for, using the username portion of the request URI. Make sure you have a matching extension for what your upstream provider is sending you, and chan_sip will respond with a 200 OK. In general, this 200 OK status code can be used for troubleshooting? Is there a log of status codes sent, or that's just done live through the console? thanks, Thufir -- _ -- 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] sipsak: 404 error
Hi, I **think** that I have user of thufir101, because I get a 200 response below, but I also get a 404. It seems to depend on how I send the ip address/fqdn? tleilax*CLI tleilax*CLI sip show users Username Secret Accountcode Def.Context ACL Forcerport 201password 201 default No Yes thufir101 password thufir101 default No Yes babyteljlfjd54545 default No Yes gs102 X58sKpZCcDfcGT0 gs102 default No Yes tleilax*CLI tleilax*CLI sip show user thufir101 * Name : thufir101 Secret : Set MD5Secret: Not set Context : default Language : en Accountcode : thufir101 AMA flags: Unknown Netborder CPD: No Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : atreides 123 ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No tleilax*CLI which would make the URI sip:thufir...@tleilax.bounceme.net ? thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thufir101@tleilax No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax message received: SIP/2.0 200 OK CSeq: 1 OPTIONS Via: SIP/2.0/UDP 127.0.1.1:52173;branch=z9hG4bK.4ca3965f;rport=52173;alias;received=192.168.1.3 User-Agent: Ekiga/4.0.1 From: sip:sipsak@127.0.1.1:52173;tag=631bb564 Call-ID: 1662760292@127.0.1.1 To: sip:thufir101@tleilax Contact: sip:thufir101@192.168.1.3 Content-Length: 0 ** reply received after 3.381 ms ** SIP/2.0 200 OK final received thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thufir...@tleilax.bounceme.net No SRV record: _sip._tcp.tleilax.bounceme.net No SRV record: _sip._udp.tleilax.bounceme.net using A record: tleilax.bounceme.net message received: SIP/2.0 200 OK CSeq: 1 OPTIONS Via: SIP/2.0/UDP 127.0.1.1:35077;branch=z9hG4bK.353a619c;rport=35077;alias;received=192.168.1.3 User-Agent: Ekiga/4.0.1 From: sip:sipsak@127.0.1.1:35077;tag=239b4596 Call-ID: 597378454@127.0.1.1 To: sip:thufir...@tleilax.bounceme.net Contact: sip:thufir101@192.168.1.3 Content-Length: 0 ** reply received after 2.987 ms ** SIP/2.0 200 OK final received thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thufir101@192.168.1.2 message received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.1.1:39721;branch=z9hG4bK.4b7b7fab;alias;received=192.168.1.3;rport=39721 From: sip:sipsak@127.0.1.1:39721;tag=6b70e831 To: sip:thufir101@192.168.1.2;tag=as34aa76ca Call-ID: 1802561585@127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.32.1-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 ** reply received after 0.665 ms ** SIP/2.0 404 Not Found final received thufir@doge:~$ thufir@doge:~$ I updated my hosts file on doge with the ip adress for tleilax...for some reason that makes it work..? any pointers, thank you, Thufir -- _ -- 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] LAN sip-to-sip
I'm reading the O'Reilly Asterisk the definitive guide, 4th ed, with a
starfish on it. In some ways, astonishing that it's not really that
definitive, it's more general -- and it only clocks in at one ream of
paper!
In any event, I'm having some port problems on my home network:
http://security.stackexchange.com/questions/81752/
I need to open ports for Asterisk to work even on a local level.
so I'm just asking in general. For SIP to SIP peer calling, and by that
I just mean ring or beep, some sort of ping, basically, just
configure the two softphones to use the IP address for the Asterisk box?
also:
tleilax:~ #
tleilax:~ # asterisk -V
Asterisk 1.8.32.1-vici
tleilax:~ #
tleilax:~ # asterisk -rm
Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
log and verbose output currently muted ('logger mute' to unmute)
Connected to Asterisk 1.8.32.1-vici currently running on tleilax (pid =
3062)
Verbosity is at least 21
tleilax*CLI
tleilax*CLI sip show peer babytel
* Name : babytel
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context : default
Subscr.Cont. : Not set
Language : en
AMA flags: Unknown
Netborder CPD: No
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
MOH Suggest : default
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid :
MaxCallBR: 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID: Yes
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : sip.babytel.ca
Addr-IP : 198.38.7.11:5060
Defaddr-IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1private
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
Status : UNREACHABLE
Useragent:
Reg. Contact :
Qualify Freq : 6 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
tleilax*CLI
tleilax*CLI sip show peers
Name/username Host Dyn Forcerport ACL Port Status
201/201 (Unspecified) D N 0UNKNOWN
babytel/1private 198.38.7.11 D N
5060 UNREACHABLE
gs102/gs102 (Unspecified) D N 0UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0
offline]
tleilax*CLI
thanks,
Thufir
--
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Re: [asterisk-users] LAN sip-to-sip
On Mon, 16 Feb 2015 16:12:04 -0500, John Novack wrote: It looks as if that is more of a question/issue with your router, rather than Asterisk. I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet wirenut You SHOULD be able to communicate between devices on the LAN without any firewall issue. I have also found with some routers that the DMZ isn't what one expects, and can get in the way, depending on the firware. Does this router have any SIP ALG setting? turn it off! As an aside, I would caution you to not have SIP 5060 exposed to the public Internet, or you will soon regret it. I am sure others will have much better information though John Novack Seems spot on. I would just add that on my LAN, it doesn't directly connect to the internet, so even an exposed 5060 port is only exposed another router. That router has firewall, etc. the netgear router connects with ethernet cable to an iogear wifi adaper. the netgear router uses DHCP and gets an IP address of 192.x.x.x from the iogear device. The iogear device gets its IP address wirelessly from the another router. That upstream router is from the ISP (has their branding), and has a firewall. So, I'm not concerned about opening ports on the netgear router :) -Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP show peers: UNREACHABLE
I'm trying to configure SIP trunking. Now, I'm referencing Asterisk
the definitive guide, 4th ed. While I don't have the page handy, I was
reading the suggestion to try SIP to SIP before proceeding to outside
connectivity. I'm aware that SIP trunking is a construct, but am,
obviously, learning the system.
What I'd like to do is from the CLI ping either the peer below, or a
peer somewhere. Unfortunately, I'm also in a double+ NAT situation at
the moment. While Skype works (mostly) from my LAN, the connection
isn't the greatest. My LAN uses a wireless bridge to connect to another
LAN. It's just a home setup; it is what it is.
How do I test a connection? How do check the settings? As far as I
can tell, the settings are correct.
tleilax:~ #
tleilax:~ # asterisk -V
Asterisk 1.8.32.1-vici
tleilax:~ #
tleilax:~ # asterisk -rm
Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
log and verbose output currently muted ('logger mute' to unmute)
Connected to Asterisk 1.8.32.1-vici currently running on tleilax (pid =
3062)
Verbosity is at least 21
tleilax*CLI
tleilax*CLI sip show peer babytel
* Name : babytel
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context : default
Subscr.Cont. : Not set
Language : en
AMA flags: Unknown
Netborder CPD: No
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
MOH Suggest : default
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid :
MaxCallBR: 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID: Yes
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : sip.babytel.ca
Addr-IP : 198.38.7.11:5060
Defaddr-IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1private
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
Status : UNREACHABLE
Useragent:
Reg. Contact :
Qualify Freq : 6 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
tleilax*CLI
tleilax*CLI sip show peers
Name/username Host Dyn Forcerport ACL Port Status
201/201 (Unspecified) D N 0UNKNOWN
babytel/1private 198.38.7.11 D
N 5060 UNREACHABLE
gs102/gs102 (Unspecified) D N 0UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0
offline]
tleilax*CLI
thanks,
Thufir
--
_
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[asterisk-users] asterisk -r spammy
when running asterisk -r, is there a way to turn off the messages? I didn't find the answer in the man page. thanks, Thufir -- _ -- 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] SugarAsterisk vs. ________
On Thu, 19 Jun 2014 17:10:48 +0100, A J Stiles wrote: The Free version of SugarCRM is no longer officially maintained; but the code can still be found if you look hard enough, and cannot easily be suppressed since the GPL lasts as long as copyright. There are also forks based on earlier Free SugarCRM versions. based on just a few minutes with the free version, it seems more than enough. I had to jump through some hoops to download it, but not too bad. There aren't as many (good) forks as I would hope for, but that's ok. Click-to-dial is pretty easy to implement. I also saw a module or plugin which did this. Darn, can't find it now. -Thufir -- _ -- 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] SugarAsterisk vs. ________
Is this completely open source? And all it's dependencies? https://github.com/trustmaster/SugarAsterisk Is it free as in free speech and free as in free beer? I know there are a few variations of SugarCRM. We're currently using an Asterisk hosted PBX with Cisco hardphones. Conceivably, we could run something like SugarAsterisk on the local network, and it would connect with the remote Asterisk? Inbound and outbound calls would be routed appropiately? Of course, it requires the Asterisk manager to be enabled... What are some alternatives? We only need the contact center software for asterisk, not asterisk itself. There's another variant (or perhaps the same thing) at: http://astercc.org/products/astercrm thanks, Thufir -- _ -- 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] SugarAsterisk vs. ________
http://www.voip-info.org/wiki/view/Asterisk+CRM+Integration lists a few options. I'm looking for, literally, the simplest FOSS CRM for click to dial functionality, but don't know where to start. thanks, Thufir -- _ -- 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] quickstart
I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir -- _ -- 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] quickstart
The headphones are Cisco phones. Ie, ext 100 and 101. I don't have the model handy at the moment. On Jun 17, 2014 2:10 PM, binary dreamer dreamer.bin...@gmail.com wrote: hi, sorry all you are asking is to have 2 internal phones call each other? the hardphones you are talking about what kind of phones are? On Tue, Jun 17, 2014 at 1:14 PM, Rainer Piper rainer.pi...@soho-piper.de wrote: Am 17.06.2014 09:04, schrieb thufir: I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir Hi ... this script will get you up and running on a debian7 distribution. code #!/bin/sh apt-get update apt-get upgrade -y asteriskversion=asterisk-12.3.2 apt-get install -y linux-headers-`uname -r` apt-get install -y build-essential apt-get install -y wget apt-get install -y libssl-dev apt-get install -y libncurses5-dev apt-get install -y libnewt-dev apt-get install -y libxml2-dev apt-get install -y libsqlite3-dev apt-get install -y libjansson-dev apt-get install -y git ln -s /usr/src/linux-headers-`uname -r` /usr/src/linux cd /usr/src ## pjsip installieren git clone https://github.com/asterisk/pjproject pjproject cd /usr/src/pjproject ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr ## um IPv6 Support in pjsip einzuschalten, muss das CFLAGS='-DPJ_HAS_IPV6=1' angegeben werden # IPV6 is turned off at default ! #./configure CFLAGS='-DPJ_HAS_IPV6=1' --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr # make dep make make install ldconfig ### check inst. # ldconfig -p | grep libpj ## System vorbereiten ## download Asterisk if [ ! -f /usr/src/$asteriskversion.tar.gz ] ; then wget http://downloads.asterisk.org/pub/telephony/asterisk/$asteriskversion.tar.gz fi if [ ! -d /usr/src/$asteriskversion ] ; then tar xvzf $asteriskversion.tar.gz fi ## erforderliche libs installieren /usr/src/$asteriskversion/contrib/scripts/install_prereq install ## optional /usr/src/$asteriskversion/contrib/scripts/get_mp3_source.sh /usr/src/$asteriskversion/contrib/scripts/get_ilbc_source.sh gcc -O2 /usr/src/$asteriskversion/contrib/utils/rawplayer.c -o /usr/bin/rawplayer ## asterisk installieren cd /usr/src/$asteriskversion ./configure make menuconfig make make install make samples make config make install-logrotate /code -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] quickstart
Pardon. My home PC is Ubuntu, 14.04. -- _ -- 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] quickstart
On Tue, 17 Jun 2014 12:14:05 +0200, Rainer Piper wrote: git clone https://github.com/asterisk/pjproject pjproject At the very least, thank you for pjsip. I'm not sure what it is yet, but seems intriguing :) I'm on Ubunutu 14.04, but will look over your script and adapt it. -Thufir -- _ -- 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] quickstart
On Tue, 17 Jun 2014 09:46:22 -0500, Rusty Newton wrote: Try following: https://wiki.asterisk.org/wiki/display/AST/Hello+World Simply use a hard phone instead of a soft-phone. Then go from there on to two phones. Perfect. I'm looking at: SIP channel driver you wanted to use, which may imply other requirements. For example if you want to use chan_pjsip... so now know to read up, in particular, on what sip channels are :) The book is just so huge, it's hard to find somewhere to start, and this looks like good place. Thank you, everyone, for the responses, I'm off to the races now. -Thufir -- _ -- 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] SQLite3 astdb back-end
How do you load the contact list? It's a database? Sqlite3? https://wiki.asterisk.org/wiki/display/AST/SQLite3+astdb+back-end I'm not clear on what this specific database does. If it's not this specific database which has contact information, which database does? thanks, Thufir -- _ -- 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] skype and SIP hardware for linux
I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under either Skype or as a SIP phone? The information I have on the driver, skypemate, is a bit sketchy. According to A-Link, the phone complies with SIP, http://www.a-link.com/us_us/IPU1.html, but the details are sketchy. No information is provided as to the interface for configuring SIP. The user manual, http://support.a-link.com/phonemate/Manual/IPU1manual_for_Linux.pdf, details using Skype but not SIP. Any user experience with this phone? For instance, has anyone used it with gizmo project or free world dialup, or even Skype? thanks, Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: skype and SIP hardware for linux
On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote: It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. Peter On 05/11/06, Thufir [EMAIL PROTECTED] wrote: I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. [...] Did I miss that info on Xlite? Sounds like this might work under linux, at least for xlite...? thanks, Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: skype and SIP hardware for linux
It seems that xlite doesn't support IAX? Too bad. While xlite does, apparently, run under linux it's not clear to me whether or not the a-link device will work with the linux version of xlite. -Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: skype and SIP hardware for linux
On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote: It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. [...] Heh, I did miss it. Yes, for windows, it specifies X-Lite software. That x-Lite isn't mentioned for Linux implies that it'll only work for windows. Curious, but not unusual, state of affairs. In any event, x-Lite doesn't support IAX, which I require. -Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: skype and SIP hardware for linux
On Sun, 05 Nov 2006 15:21:24 +0200, Dovid B wrote: I downloaded a softphone called kiax last night. Its working great. I was real tired then so I dont remember where I got it from. Hope that helps. (and its open source as well as they give you the source files for it :) ) [...] http://www.kiax.org/screenshots/ looks good. I'm looking at http://www.nslu2-linux.org/wiki/HowTo/ConnectUSBPhone, which _appears_ to describe the same phone. If so, this brings me full circle to asterisk as a solution. I'd definitely need the IAX, which kiax supports. Are the nslu2 folks describing hacking the http://www.yealink.com/english/prodetail_p1k.htm phone, or using that phone _with_ a slug? If I can run asterisk on my computer, and not hack any hardware, that'd be preferable. thanks, Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: PAP2 to use on my asterisk.
On Fri, 03 Nov 2006 09:26:22 +0100, twanny wrote: Hello, I nearly forgot about this mailing list! I accidentally bought a vonage enabled PAP2 to use on my asterisk, however it's locked and I have no access to the admin password. Anyone unlocked it before? Please send procedure. Make a cc to my address please. Thanks a million. [...] http://www.grandstream.com/y-286.htm Does everything which the vonage PAP2 does (Do you mean linksys?). The password for users is 123 or, for admin, admin. Even works under linux, as it responds to HTTP POST requests. No driver necesarry. They also make phones, or adapters with multiple jacks for phones, if you need more than one jack. -Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is IAX required for firewall and router?
I'm trying to understand IAX and whether or not it would solve my difficulties: 'The primary goals for IAX were to minimize bandwidth used in media transmissions, with particular attention drawn to control and individual voice calls, and to provide native support for NAT (Network Address Translation) transparency. Another goal is to be easy to use behind firewalls.' http://en.wikipedia.org/wiki/IAX This sounds good, as I have a NAT traversal problem, as well as others. I'm not the system admin for my network, and am behind a router and firewall. Skype works ok as a softphone, but I'd rather something like http://www.grandstream.com/y-286.htm with which I can use a real telephone. Either an ATA or a SIP phone, I'm not sure yet. In any event, there's no point in buying the hardware until I get the networking sorted. Somehow I need to TUNNEL throught the router and firewall? I was looking at http://en.wikipedia.org/wiki/STUN as one way of doing this. The Grandstream FAQ explains how to do this: '5. How do I setup my Grandstream Phone for go2call network? typical configuration is: SIP Server: voip01.go2call.com Outbound proxy: (Should leave it blank, because it's a GW) User ID: x (your Go2Call PIN number) Authentication ID: same as your User ID Password: xxx (Your Go2Call password) NAT Traversal: YES (WITHOUT setting the STUN server) 6. How do I setup my Grandstream Phone for FWD service? typical configuration is: SIP Server: fwd.pulver.com outbound proxy: 192.246.69.247:5082 (used only when behind firewall, otherwise leave it blank) User ID: xx (your FWD account number) Authentication/Login ID: x (same as above, your FWD account number) Password: x (your FWD password) NAT Traversal: No (You need to set up your STUN server if you don't have outbound proxy)' http://www.grandstream.com/FAQ.pdf However, most everything I read on the subject starts with the assumption that there's physical and administrative access to the router. Neither of those hold for me :( I would like to set this up prior to purchasing the hardware, but I don't even know that it's possible. Again, I don't have physical access to the router, so the ATA would have to connect through the computer, which connects to the router. It's not possible to directly connect any hardware to the router. No, it's not possible to use a switch. thanks, Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
