Quantcast
Channel: Asterisk Forums
Viewing all 7460 articles
Browse latest View live

Asterisk 13 configuration for Outbound trunk

$
0
0
Hi, I am just playing around with the new Asterisk 13 version and two outbound endpoints.
The configuration for receiving calls from the first outbound endpoint is working but I have problems to set up the outbound endpoint which receives the calls from me which will go to the public telephone network. Those two endpoints have different ip addresses!
The outbound server doesn't need auth settings, it will recognize the asterisk 13 server based on it's IP address in the INVITE section.
After many configuration and dialplan changes I am totally frustrated and can't see what I am doing wrong and the documentation for Asterisk 13 is not really available. Therefore it is also not helpful when I post my configuration settings.
So what would theoretically be the basic dialplan setting and pjsip configuration to get my setting working? (Or where can I find better documentation?)

Statistics : Posted by niels14 • on Wed Dec 03, 2014 6:31 am • Replies 1 • Views 74

3 RE-INVITES send for asterisk

$
0
0
;----------------------------------------- NAT SUPPORT ------------------------
;
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
;
; When Asterisk is behind a NAT device, the "local" address (and port) that
; a socket is bound to has different values when seen from the inside or
; from the outside of the NATted network. Unfortunately this address must
; be communicated to the outside (e.g. in SIP and SDP messages), and in
; order to determine the correct value Asterisk needs to know:
;
; + whether it is talking to someone "inside" or "outside" of the NATted network.
; This is configured by assigning the "localnet" parameter with a list
; of network addresses that are considered "inside" of the NATted network.
; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
; Multiple entries are allowed, e.g. a reasonable set is the following:
;
; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
;
; + the "externally visible" address and port number to be used when talking
; to a host outside the NAT. This information is derived by one of the
; following (mutually exclusive) config file parameters:
;
; a. "externaddr = hostname[:port]" specifies a static address[:port] to
; be used in SIP and SDP messages.
; The hostname is looked up only once, when [re]loading sip.conf .
; If a port number is not present, use the port specified in the "udpbindaddr"
; (which is not guaranteed to work correctly, because a NAT box might remap the
; port number as well as the address).
; This approach can be useful if you have a NAT device where you can
; configure the mapping statically. Examples:
;
; externaddr = 12.34.56.78 ; use this address.
; externaddr = 12.34.56.78:9900 ; use this address and port.
; externaddr = mynat.my.org:12600 ; Public address of my nat box.
; externtcpport = 9900 ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT.
; ; externtcpport will default to the externaddr or externhost port if either one is set.
; externtlsport = 12600 ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
; ; externtlsport port will default to the RFC designated port of 5061.
;
; b. "externhost = hostname[:port]" is similar to "externaddr" except
; that the hostname is looked up every "externrefresh" seconds
; (default 10s). This can be useful when your NAT device lets you choose
; the port mapping, but the IP address is dynamic.
; Beware, you might suffer from service disruption when the name server
; resolution fails. Examples:
;
; externhost=foo.dyndns.net ; refreshed periodically
; externrefresh=180 ; change the refresh interval
;
; Note that at the moment all these mechanism work only for the SIP socket.
; The IP address discovered with externaddr/externhost is reused for
; media sessions as well, but the port numbers are not remapped so you
; may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the "externaddr" and
; "externhost" might not help you configure addresses properly.
;
; NOTE 2: when using "externaddr" or "externhost", the address part is
; also used as the external address for media sessions. Thus, the port
; information in the SDP may be wrong!
;
; In addition to the above, Asterisk has an additional "nat" parameter to
; address NAT-related issues in incoming SIP or media sessions.
; In particular, depending on the 'nat= ' settings described below, Asterisk
; may override the address/port information specified in the SIP/SDP messages,
; and use the information (sender address) supplied by the network stack instead.
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
; nat = no ; Do no special NAT handling other than RFC3581
; nat = force_rport ; Pretend there was an rport parameter even if there wasn't
; nat = comedia ; Send media to the port Asterisk received it from regardless
; ; of where the SDP says to send it.
; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
;
; The nat settings can be combined. For example, to set both force_rport and comedia
; one would set nat=force_rport,comedia. If any of the comma-separated options is 'no',
; Asterisk will ignore any other settings and set nat=no. If one of the "auto" settings
; is used in conjunction with its non-auto counterpart (nat=comedia,auto_comedia), then
; the non-auto option will be ignored.
;
; The RFC 3581-defined 'rport' parameter allows a client to request that Asterisk send
; SIP responses to it via the source IP and port from which the request originated
; instead of the address/port listed in the top-most Via header. This is useful if a
; client knows that it is behind a NAT and therefore cannot guess from what address/port
; its request will be sent. Asterisk will always honor the 'rport' parameter if it is
; sent. The force_rport setting causes Asterisk to always send responses back to the
; address/port from which it received requests; even if the other side doesn't support
; adding the 'rport' parameter.
;
; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
; draft form. This method is used to accomodate endpoints that may be located behind
; NAT devices, and as such the address/port they tell Asterisk to send RTP packets to
; for their media streams is not the actual address/port that will be used on the nearer
; side of the NAT.
;
; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
; the nat setting in a peer definition, then the peer username will be discoverable
; by outside parties as Asterisk will respond to different ports for defined and
; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
; other, then valid peers with settings differing from those in the general section will
; be discoverable.
;
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
; to receive them on.
;
; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
; the media_address configuration option. This is only applicable to the general section and
; can not be set per-user or per-peer.
;
; media_address = 172.16.42.1
;
; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
; perceived external network address has changed. When the stun_monitor is installed and
; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
; of network change has occurred. By default this option is enabled, but only takes effect once
; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
; generate all outbound registrations on a network change, use the option below to disable
; this feature.
;
; subscribe_network_change_event = yes ; on by default
;
; ICE/STUN/TURN usage can be disabled globally or on a per-peer basis using the icesupport
; configuration option. When set to yes ICE support is enabled. When set to no it is disabled.
;
; icesupport = no

is nothing, dont work, i need to edit header SDP, plz tell me how

Statistics : Posted by javierr_vv • on Mon Dec 01, 2014 4:49 pm • Replies 5 • Views 223

3 RE-INVITES send for asterisk

$
0
0
Look for externip,localnet in your sip.conf.

--Satish Barot

Statistics : Posted by javierr_vv • on Mon Dec 01, 2014 4:49 pm • Replies 5 • Views 223

3 RE-INVITES send for asterisk

$
0
0
There is a lot of information on the web about using Asterisk with NAT. There is also quite a bit in the sample configuration file.

Statistics : Posted by javierr_vv • on Mon Dec 01, 2014 4:49 pm • Replies 5 • Views 223

3 RE-INVITES send for asterisk

$
0
0
sorry for put the topic here, but i need tu edit SDP, i need put ip public, how do it?, in fact the second invite is good, but with third came dont work!

Statistics : Posted by javierr_vv • on Mon Dec 01, 2014 4:49 pm • Replies 5 • Views 223

3 RE-INVITES send for asterisk

$
0
0
You tell Asterisk it''s public address, so that it is aware that it is using NAT.

Typically the first re-invite is enabling direct media and the second re-invite is cancelling it, because the other party has hung up.

In futures, please don't use discussion forums for support questions.

Statistics : Posted by javierr_vv • on Mon Dec 01, 2014 4:49 pm • Replies 5 • Views 223

3 RE-INVITES send for asterisk

$
0
0
UA1-----Router 1 NAT-------Asterisk-------Router 2 NAT----------- UA 2

Image
when try to establish point-to-point communication, there are 3 packages invite
SDP first:
Code: v=0
o=root 639080599 639080600 IN IP4 192.168.1.3
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.3
t=0 0
m=audio 33214 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


SDP second:
Code: v=0
o=root 639080599 639080601 IN IP4 152.XXX.XXX.XXX (ip public)
s=Asterisk PBX 11.14.1
c=IN IP4 152.74.21.54
t=0 0
m=audio 33214 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


SDP third

Code: v=0
o=root 639080599 639080602 IN IP4 192.168.1.3
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.3
t=0 0
m=audio 33214 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

between the second and third package SDP, works smoothly, as it's sent to the correct public IP and port, but then asterisk send third package INVITE then send private ip, so the router drops the packet, how do I configure asterisk and not send the third package is bothering.
regards!

Statistics : Posted by javierr_vv • on Mon Dec 01, 2014 4:49 pm • Replies 5 • Views 223

New iOS Digium Softphone app

$
0
0
I removed my extension from my D70 phone and created a Softphone assignment. After I got the generated email on my iPhone and clicked the link and entered my extension login info, the Softphone app opened but just said "Configuring" forever. I killed the app and tried a few more times but no luck.

Statistics : Posted by bberryman • on Wed Dec 03, 2014 1:16 pm • Replies 1 • Views 45

New iOS Digium Softphone app

Iphone app

$
0
0
They just sent out an announcement email detailing the upcoming app refresh, and feature set, so it seems they're aware of the existing issues and estimate release sometime in October.

Statistics : Posted by rnewberger • on Tue Aug 12, 2014 6:27 pm • Replies 9 • Views 983

Iphone app

$
0
0
I can confirm, my Switchvox mobile app has worked through all iterations of iOS in the past, but is dead now. The app will open, but I cannot get any new data in it (it only has old data from prior to me upgrading), I cannot make calls through it, nor use anything in it without it hard-locking.

I'm going to open a ticket with technical support, to ensure they know about it and if it truly is an issue, hopefully we can get it on Switchvox Today.

Statistics : Posted by rnewberger • on Tue Aug 12, 2014 6:27 pm • Replies 9 • Views 983

Iphone app

$
0
0
Huh, I guess my last reply was blocked complaining about Digium's poor app support. No, it doesn't work on iOS 8 and that's a shame.

Statistics : Posted by rnewberger • on Tue Aug 12, 2014 6:27 pm • Replies 9 • Views 983

Iphone app

$
0
0
No official news on the new app yet. I haven't tried it on 8.0 yet, no access to it.

There was a brief time where the old iOS app didn't work, but it didn't have anything to do with iOS 7, it had to do with a Switchvox update and there has since been a fix on the back-end for it.

Statistics : Posted by rnewberger • on Tue Aug 12, 2014 6:27 pm • Replies 9 • Views 983

Iphone app

$
0
0
Rich,

When I still was using my iPhone a few months ago before switching to the HTC One M8, I had issues with the Switchvox app crashing on iOS 7. Read this thread to see the workaround. An app update is long overdue for sure.

http://forums.asterisk.org/viewtopic.php?f=38&t=89710

Bob

Statistics : Posted by rnewberger • on Tue Aug 12, 2014 6:27 pm • Replies 9 • Views 983

Iphone app

$
0
0
Any Developers on here that are running IOS Version 8.0? Is switchvox working for you?
Anyone else what version of IOS are you on and is switchvox app working for you.
Ive not been able to use it for a few versions Switchvox server is up to date.

Im going to see if I can get the logs off my phone and see where this thing is bombing.

Any word on updates to the app?

Thanks,
Rich

Statistics : Posted by rnewberger • on Tue Aug 12, 2014 6:27 pm • Replies 9 • Views 983

Direct Incoming Call To Extension Based On Spoken Request

$
0
0
If a new incoming call exceeds the maximum queue size (see Enabling the voice channel and configuring the general settings in the Zendesk Administrator Guide), the total number of calls that are allowed to wait in the queue, the caller is sent to voicemail.
If a new incoming call is within the limits of the maximum queue size, the caller is queued for the next available agent. If you've configured that number to be routed to a specific group or set of groups, the caller must wait for an agent in that group to be available. The caller can dial 1 at anytime to leave a voicemail.
Software companies in ghaziabad
Ghaziabad Software Companies
IT Company in Ghaziabad

Statistics : Posted by OOsorio • on Thu Nov 20, 2014 2:20 pm • Replies 3 • Views 231

Direct Incoming Call To Extension Based On Spoken Request

$
0
0
Thank you,

Statistics : Posted by OOsorio • on Thu Nov 20, 2014 2:20 pm • Replies 3 • Views 231

Direct Incoming Call To Extension Based On Spoken Request

$
0
0
Not natively with Switchvox, in any current release, but we've used this product before:
http://www.incendonet.com/speechbridge-smb/

It's relatively inexpensive, connects to Switchvox over SIP and does exactly what you're looking for.

Statistics : Posted by OOsorio • on Thu Nov 20, 2014 2:20 pm • Replies 3 • Views 231

Direct Incoming Call To Extension Based On Spoken Request

$
0
0
I would like to have the IVR ask the caller for the contact name and, based on speech recognition, direct the call to that contacts extension. Can this be done?

Statistics : Posted by OOsorio • on Thu Nov 20, 2014 2:20 pm • Replies 3 • Views 231

[SOLVED] Corrupt .gsm files from MixMonitor?

$
0
0
Hi

Thanks for the reply!

Turns out my upstream provider had me connected to an experimental server on their side that they were using for random number generation testing.

It appears they increased the G729 payload to 70ms instead of the standard 20ms that they always use on their regular servers.

As far as I can determine this was what was causing the corrupt .gsm recordings. I got the IP of one of their regular production servers, edited sip.conf to register to that server instead of the experimental one, and the box is now running fine.

I did replace the entire server as well, but I think the problem was somehow caused by their testing server to which I was unwittingly SIP subscribed to....

I learned something again - it appears that if a trunk you're sending and receiving G729 encoded voice / RTP on has a widely differing payload size from what Asterisk 1.8.11.0 is set to expect (I explicitly specify allow=G729:20 in my sip.conf for this connection) the audio itself may be fine bi-directionally but .gsm mixmonitored recordings of same audio will be corrupt...?

Anyway, thanks for taking the time to respond, much obliged!

Statistics : Posted by rylan76 • on Wed Dec 03, 2014 7:53 am • Replies 2 • Views 160
Viewing all 7460 articles
Browse latest View live


<script src="https://jsc.adskeeper.com/r/s/rssing.com.1596347.js" async> </script>