similar to: One way audio on outgoing calls

Displaying 20 results from an estimated 6000 matches similar to: "One way audio on outgoing calls"

2023 Jun 21
1
Asterisk not replacing private FROM ip with public IP in INVITE
You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:
2017 Oct 09
6
PJSIP, NAT and STUN/ICE
I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's
2020 Sep 08
3
Some calls drop after 30 seconds
    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:
2023 Apr 08
1
TLS and NAT
Hello Steve, use the following configuration for the transport and bind this transport to the trunk: [transport_name] type=transport protocol=tls bind=192.168.13.24 ; your bind IP ca_list_file=/etc/pki/tls/certs/ca-bundle.crt ; method=tlsv1_2 verify_server=yes allow_reload=no ;tos=0xb8 ;cos=3 external_media_address=your.ext.host.name ; hostname pointing to your ext. IP
2023 Apr 09
1
TLS and NAT
Thanks, Michael. A few questions: Is [transport_name] a reserved word, or am I supposed to replace it with a name of my own, like '[did-transport]'? Some of the keywords I haven't seen before. Is ca_list_file supposed to be an aggregate of the public and private key? And what are the 'method,' 'tos' and 'cos' keywords, which are commented out in your
2015 Mar 05
2
PJSIP configuration for AWS/EC2 based Asterisk 13.1.0
Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and "see" them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration
2016 Sep 14
2
Asterisk 13 externip
Hi, What is the equal option for externip in asterisk 13 with pjsip. I have tried external_media_address=XX.XX.XX.XX external_signaling_address=XX.XX.XX.XX but asterisk 13 writes local ip to the from header. any suggestions? Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL:
2023 Apr 07
1
TLS and NAT
I want to configure communication with my phone provider using TLS for all the obvious reasons. Since I'm behind a firewall, I'll be needing to do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but none for TLS plus NAT. Would it be correct to set up the TLS transport stanza to look like the [transport-udp-nat] stanza example, replacing UDP with TLS in lines like
2016 Sep 15
2
Asterisk 13 externip
On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> Hi, >> >> What is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >>
2016 Sep 15
2
Asterisk 13 externip
On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga <mgliyanage.rc at gmail.com > wrote: > Hi, > > Thanks for the reply. > > Yes my PABX is on the cloud when I try to register to the server, the > server sends registration OK with public address but OPTION method > includes the private address of the server in from header not the public > address. I have include
2016 Mar 03
3
RTP / NAT question ( pjsip )
Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP: 17:07:57.130212 IP
2023 Aug 18
2
PJSIP Losing knowledge of external_media_address
I've seen this happen three times in the wild now.  I've been trying to isolate the source of the issue, but so far it seems like there's not enough debug output to know why this occurs. Long story short: - Start Asterisk - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind NAT).  SIP is handled correctly, Asterisk responds OK with RTP media address of
2023 Jul 25
1
Can ShanSpy be used on Local Channels?
    Does anyone know if Chanspy can be used with local channels? Since agents on queues need to use local channels like Local/XXXX at from-queue/n to make sure that all of their registered extensions ring we are now having a problem trying to use ChanSpy to listen to calls.  Since we do not know if the agent is on their desk phone or a softphone (which use different identifiers) we cannot set
2016 Jan 26
2
PJSIP Stun/ICE
I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack). How do we implement STUN/ICE on the server behind the dynamic Address. It does not appear to be registering properly without knowing the NAT pubic
2023 Jun 21
2
Asterisk not replacing private FROM ip with public IP in INVITE
I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction: From: "MYNAME" <sip:16667778888 at 172.31.253.4>;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 The IP address above is an internal/non-routable IP, so Twilio is rejecting it. For some
2019 Nov 01
2
Stuck "channel"
I have tried both by hand and hitting tab to auto complete: *CLI> channel request hangup Message/ast_msg_queue Message/ast_msg_queue is not a known channel On 31/10/19 14:18, Sean Bright wrote: > On 10/31/2019 2:13 PM, Carlos Chavez wrote: >> I assume this is something created by Freepbx.  If I do a "channel >> request hangup" it tells me the channel does not exist.
2023 Jun 21
1
PJSIP not performing outbound authentication
    Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls
2020 Oct 02
1
PJSIP_DIAL_CONTACTS and Queues
    Is there a solution to dial multiple contacts for a Queue agent?  Since the pandemic started many of our customers have begun to move agents off site.  Since most of them were using softphones we did not have any problems but now we have one case where the agents have a desk phone in their cubicle and are using a softphone from home.  For regular calls there is no problem as
2017 Jul 29
2
[asterisk13] Multiple transport objects of same protocol in pjsip.conf
Scenario: Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisk's wiki, the transport section of pjsip.conf is configured as follows: ; Transport via UDP [transport-nat-udp] type= transport
2014 Dec 16
3
PJSIP configuration question
Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*