[asterisk-users] DTMF behavior in asterisk 12 with PJSIP
Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you know is there is a plan to develop it? 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk does not transcode the DTMF signals, therefore DTMF is not working. It used to work on release 11. This is really bad. Do you know of a solution to this issue? Maybe some settings? Thanks, Yaron. -- _ -- 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] Port number in From URI on Asterisk 12 PJSIP
Hello, I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM URI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIP servers (it doesn't recognize the response / request). Below you can see an incoming INVITE and the outgoing 200OK response. I have highlighted the issue in Yellow. Does anyone know of a solution / workaround for this issue? --- Received SIP request (648 bytes) from UDP:172.16.60.160:5061 --- INVITE sip:039988120F@172.16.60.160:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK-29450-3-0 Max-Forwards: 60 From: sip:39937841@192.168.225.2:5061;user=phone;tag=3 To: sip:039988120F@172.16.60.160:5060;user=phone Call-ID: 3-29450@172.16.60.160 CSeq: 1 INVITE Contact: sip:10.1.1.1:5060 User-Agent: Simulator Supported: 100rel Privacy: id Min-SE: 90 Content-Type: application/sdp Content-Length: 201 v=0 o=172.16.60.160 10864 2 IN IP4 172.16.60.160 s=SIP Call c=IN IP4 172.16.60.160 t=0 0 a=sendrecv m=audio 6 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- Transmitting SIP response (730 bytes) to UDP:172.16.60.160:5061 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1:5060 ;rport;received=172.16.60.160;branch=z9hG4bK-29450-3-0 Call-ID: 3-29450@172.16.60.160 From: sip:39937841@192.168.225.2;user=phone;tag=3 To: sip:039988120F@172.16.60.160 ;user=phone;tag=4f7ef94f-fb15-4bf5-94bd-4e43fe-299655 CSeq: 1 INVITE Contact: sip:172.16.60.160:5060 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 189 v=0 o=- 10864 4 IN IP4 10.2.0.67 s=Asterisk c=IN IP4 172.16.60.160 t=0 0 m=audio 19404 RTP/AVP 8 c=IN IP4 172.16.60.160 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:150 a=sendrecv -- _ -- 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] Voicemail ODBC Storage
On Saturday, October 25, 2014 09:09:57 PM Dan Journo wrote: Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? It may cause you trouble when using PostgreSQL: https://issues.asterisk.org/jira/browse/ASTERISK-24441 -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Kamailio Load Balancing
I have a question please...i want to use Kamailio to do load balancing between multiple asterisk servers - say two servers - . I suppose that the configuration of the SIP users should be identical on the two servers but i do not know what is the best way to achieve this whether it will be by configuring Mysql replication between the two servers or by configuring one of the server's database as a shared database ?! Please can u explain that point ? Best Regards -- _ -- 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] DTMF behavior in asterisk 12 with PJSIP
On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you know is there is a plan to develop it? No one that I'm aware of is currently working on that. As Asterisk is an open source project, if having the 'auto' feature added to the PJSIP stack is something you're interested in, you should consider writing a patch for the project [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk does not transcode the DTMF signals, therefore DTMF is not working. It used to work on release 11. This is really bad. Do you know of a solution to this issue? Maybe some settings? That actually is a bug. You are most likely ending up in a native packet to packet bridge (or a native remote bridge), which does not decode the RTP stream. Hence, the inband DTMF or RFC 2833 DTMF is not being decoded and is being passed to the other side. Please do open an issue for that [2]. Make sure you provide a full DEBUG log, as that will illustrate what is actually occurring. Note that you can work around that issue by adding a feature flag to whatever application caused the bridging to occur. [2] https://issues.asterisk.org/jira -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 11.13.1: unable to load sip.conf (or iax )
On 10/24/2014 03:24 PM, Jeffrey Ollie wrote: On Fri, Oct 24, 2014 at 1:47 PM, sean darcy seandar...@gmail.com wrote: On 10/24/2014 02:21 PM, Jeffrey Ollie wrote: restorecon -rv /etc/asterisk I'd never have guessed. Yeah, if you mv the data instead of cp the data from one place to the other, the SElinux labels don't get updated. I like SElinux, but it would be nice if there were better error messages... Although, if you're on Fedora 20, this is a pretty good description of what was going wrong and how to diagnose/solve it: http://danwalsh.livejournal.com/65777.html Thanks for the reference, but in my case systemctl status asterisk, which I did try, had no reference to selinux. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting Music on Hold with the Manager Interface
Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within the same login, socket session. Just trying to add this additional task. This is from PHP as you may have recognized. I have also tried surrounding musicclass with CHANNEL() but that didn't work and didn't seem right anyhow since it already knows it's a channel variable. Thanks in advance for any help on this.# Set the Music on Hold fputs($socket2, Action: Setvar\r\n); fputs($socket2, Channel: .$channel.\r\n); fputs($socket2, Variable: musicclass\r\n); fputs($socket2, Value: .$mohclass.\r\n); -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
