similar to: Asterisk 11.5 not honoring RTP port change in RE-INVITE

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value "a=sendrecv" is present, according to the rfc3264 the sdp value a must be mark with "sendonly". I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and

Hi folks, in order to establish backward compatibility we made an image that automatically detects if the other side does not support RFC3264. Please try it out, we would be very interested if this image is a progress! http://snom.com/download/share/snom200-2.03b-SIP.bin Thanks, CS

Hi I'm trying to get the spa 8000 used with a fax machine using t38 passthru i have tried with 1.6.2 and 1.8.3 and is still a no go the provider i use is 012 in israel wich supports t38 (i use it with ffa) could anybody give me a clue how to get this working if it should t38pt is set to yes in sip.conf Thanks, Israel -------------- next part -------------- An HTML attachment was

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

I hope this isn''t too far off topic. I did a little bit of pricing homework a few months ago on new T1 cards. The idea was to build a Linux based router/firewall. After all, Ethernet NICs are easily available for less than $10 today. But all the T1 cards I found cost a fortune. And about a year ago, I learned the list price for a complete Adtran Netvanta 3200 router with T1 card is

> Page 3: > > To be compliant with this specification, implementations MUST support > 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. > The sampling rate MUST be 8, 16 or 32 kHz. > > There is a type above after (narrowband), there is a " extra character. > > I don't understand what is the motivation to specify "SHOULD

B.H. Hello, all I'm using Asterisk 11.7, connected to PSTN using SIP trunk. I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from. I have googled for a solution and found this patch:

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

Greetings All, We've been working hard on new codec negotiation stuff for Asterisk 18 and we've got some stuff to run by you. It's a lot so please read carefully. To give you some idea of just how difficult a job this is, a simple call from Alice to Bob currently causes 8 attempts to reconcile codecs between them in app_dial, chan_pjsip, res_pjsip_session and res_pjsip_sdp_rtp. If

Hi... Out of sudden our samba file servers are no longer honoring secondary group membership. I appears that the fileservers are no longer seeing groups. When I do a 'net -U <domUser> rpc group list' on the PDC everything is fine. I can see all groups. When I do this on the fileservers I do not receive anything. We operate samba 4.3.9 both as classic PDC (for a couple of reason

Hi Thomas, >----Messaggio originale---- >Da: thomas@kupper.org >Data: 22/01/2010 10.48 >A: "Leszek Ciesielski"<skolima@gmail.com> >Cc: "linux-btrfs"<linux-btrfs@vger.kernel.org> >Ogg: Re: mount command not honoring rootflags passed > > >On 22 Jan 2010, at 10:40, Leszek Ciesielski wrote: > >> On Fri, Jan 22, 2010 at 10:30 AM,

One our servers at work, I use ssh/rdist to sync binaries on various machines...one master server, several client machines. This requires the ability to ssh to the remote host without a password, so I have a .rhosts setup on the remote host, trusting the 'central server'... With pre25, I can't seem to get this to work anymore... I have PermitRootLogin yes IgnoreRhosts no set ...

Running 11.10.2 on NetBSD 6.1.4 but I observed this on 11.11.0 as well. I have a directory which, through a combination of NULL mount and UNION mount contains everything in the installed config directory /usr/pkg/etc/asterisk except for my modified versions of those files. Basically I mount_null /usr/pkg/etc/asterisk on /usr/local/etc/asterisk and then mount_union my SVN directory with my

On 08/07/2015 09:53 AM, Silver Sky Soft Services, Inc. wrote: > Hi, > I'm trying to create a snapshot on a specific disk by using diskspec > option. However, libvirt/virsh seems to ignore it and takes a snapshot of > all disks. Am I using this option incorrectly? > > In below examples, I'm trying to snapshot only vdb. However even vda is > getting snapshotted

I'm running Samba 4.0.7 on CentOS 6.4 as a AD DC with s3fs. I have a shared directory with the setgid bit set. From the shell on the server, new files and directories inherit the group as expected. However, new items created through samba get the user's primary group instead. Config for the share is super simple: [test] path = /srv/test read only = no Sounds like a