- FreeSWITCH-users - FreeSWITCH Mailing Lists

tty support
by Jason Holden 11 Feb '15

11 Feb '15

Can anyone provide any guidance on integrating support for TTY machines with Freeswitch?

2 1

GXP2160 no audio in or out
by Paul Atreides 11 Feb '15

11 Feb '15

Hi, when I call a Grandstream GXP2160 and pick up the call then I can't hear anything. Nothing is going in or out. Its just silence. Registration works, calling and receiving calls works too. Other Phones work. xlite and snom D725 Does anyone know? Thank you

2 2

Bad git clone URL in stash
by Steven Ayre 11 Feb '15

11 Feb '15

Does anyone know why https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse provides the URL 'git://freeswitch.org/fs/freeswitch.git' when you click Clone? AFAIK that's the old URL that's no longer functional (connection times out) and should be 'https://freeswitch.org/stash/scm/fs/freeswitch.git' Steve

3 4

How can I trigger a ZRTP connection on an existing call?
by Peter Steinbach 11 Feb '15

11 Feb '15

Hello, I want to establish ZRTP, after I am bridging 2 calls, which have been active already before. My scenario is as follows: * Caller A is calling Freeswitch, the call is answered and an announcement is played * In the meantime Freeswitch calls another Party B and connects it to an internal IVR, where also announcements are played. * Now, after a while, I decide to connect the calls together, then send an "uuid_media off <uuid>" to the calling party A leg, reinvites occur and bypass media works. * However the 2 SIP clients (Csipsimple) do not start a ZRTP session * If caller A calls B directly via bypass media, the ZRTP is active, by the way. I suspect that ZRTP somehow should be forced or triggered. Is there a way to do that? I already invited the B leg with "zrtp_secure_media=true" and issued uuid_setvar 1bd508af-19c7-4e38-8621-731baaf33920 zrtp_secure_media true uuid_media off 1bd508af-19c7-4e38-8621-731baaf33920 to the A leg. But this did not work. The main question should be: How can I trigger a ZRTP connection on an existing call? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

2 4

Cent OS 6.x
by John Nash 11 Feb '15

11 Feb '15

As per documentation and old posts, we should use cent OS 5.X ?..is that still valid or latest cent OS 7 solves that issue?

2 1

Fine tuning Opus in dialplan
by Emrah 11 Feb '15

11 Feb '15

Hi, I use Opus even for narrowband calls because of all its bells and whistles that PCM doesn't have. The trouble is that I always end up negotiating a crazy sampling rate, more demanding in every way and not necessary for a basic narrowband call. I know this because if I record the call, one leg is recording a full 48kHz audio that is unbelievable and the other one is a regular 8kHz sad PSTN call. I have 40kHz going to waste here. I see issue 6947 seems to be talking about the same thing and was wondering if something had been done about it. https://freeswitch.org/jira/browse/FS-6947 It shows it's been solved, but I cannot see any dialplan variables I could set to fine tune my Opus codec negotiation.

3 4

Notify after forwarded Call is finished
by Raphael Lechner 11 Feb '15

11 Feb '15

Hi, I’m trying to find out a special solution for our customer. He needs to understand when the forwarded call is finished. The workflow is the following: - User A is calling a external number (User B) and after the Caller has answered the call,he puts the Call on Hold - User A is calling the second external number (User C) (with the BLF keys) and says that on the other line is person X and do an Attended Transfer And the problem is that the User A want to know when the call is finished for calling the next person. Is there a way to understand when the call is finished? It it possible to use the BLF functions for that? I tried to complete that using the dial plan with hangup_after_bridge=false and continue_on_fail=true for calling a command something after the bridge but still without luck and the BLF key for User C is also always Green. Plan B would be to parse the channels table and do send a message to the yealink phones after the call has left the channels table… We are using FS 1.4.14, Yealink Phones and a patton smartnode as ISDN/SIP Gateway. Any hint? Maybe to watch a special Event to understand when the call is finished? Thank you very much! Raphael

2 3

SIP Proxy - Kamailio/Opens* load-balancing - simple example please?
by Avi Marcus 10 Feb '15

10 Feb '15

Hi - I know this has come up several times and I see several examples even on the old wiki - but I've never used a SIP proxy before, and I don't understand most of the configuration in the examples. Here's what I believe is a fairly common use case: 1) I have inbound calls via various carriers. Some only support an IP endpoint (so DNS won't work) and some don't support 302 redirects, so I can't use a stateless FS endpoint either, so I need a sip proxy. 2) I need a SIP proxy that will route calls to various FreeSWITCH endpoints. (Preferably, it should route registration too (or manage registration list itself?)) 3) I need the ability to pull freeswitch nodes out of the routing (waiting for them to drain is fine, I don't need failover of live calls) to perform maintenance, and then add them back to the routing. 4) I should be able to have this proxy on a floating IP that I can move *this *too, *without downtime*, for maintenance work. I think this is mostly handled by the <https://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeS…> examples <http://wiki.freeswitch.org/wiki/SBC_Setup> I <https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> found <http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegrati…> - but the opensips configuration files are hundreds of lines that I don't understand. If it's just routing calls to a backend, shouldn't that be possible in a small numbers of lines that are more understandable? I imagine someone can probably just point me to a tutorial/working code that I can use. Several have been shared but I don't recall any one being particularly simple... Also, most of them tell you about compiling code. FS has been released in packages - has opensips? That might cut off many steps from an updated tutorial. Thanks! -Avi Marcus

11 14

Adding P-Charge-Info SIP Header in a SIP Profile
by Ali Pey 10 Feb '15

10 Feb '15

Hello, Is there a way to add the P-Charge-Info header in a SIP Profile? So all outgoing Invites from that sip profile will always include the P-Charge_Info header. Thanks, Ali Pey

3 6

DNS-based dialing?
by Thomas Auge 10 Feb '15

10 Feb '15

Maybe I'm missing the obvious, but I can't find a way to make sofia just contact the host part of a sip address (or use the corresponding NAPTR/SRV records) without creating a gateway record for each domain/server. A pointer in the right direction would be greatly appreciated. :-)

2 2