[HELP] multi channel sip carrier

[ken101]

Hi,

Im new here. and also learning Vicidial. I just got my sip provider to give us multi channel sip carrier. I want to be able to dial 5 agent simultaniously, with these carrier. The problem is, when i dial with 2nd agent, the one we calling cannot hear 2nd agents voice. 1st agent is okay.

Here is my carrier setting:

register => 100-xxxxxx:xxxxxxx@xxx.xxx.xxx:5060

Account Entry:

[Test-carrier]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
secret=Vxxxxxx
username=100-xxx
host=1xxxxxxxx
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes

Dialplan Entry:


exten => _63X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _63X.,2,Dial(SIP/${EXTEN}@Test-carrier,55,o)
exten => _63X.,3,Hangup

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Are these autodialed calls?

4) Is there a difference in the asterisk CLI output between the two calls?

5) Have you checked the SIP debug output for one of these "second calls" to see if there is a problem with the handshake?

6) Is the server virtual?

7) Describe your network configuration: Does the Vicidial server have its own public IP or is it sharing with other devices on a private network?

[ken101]

- ViciBox v.7.0.4-170113 from ViciBox_v.7.x86_64-7.0.4.iso | Asterisk 11.25.1-vici built by abuild @ cloud107 on a x86_64 running Linux on 2017-01-08 07:00:49 UTC |Single Server| No Digium/Sangoma Hardware | No Extra Software After Installation | asrock | AMD A6-6400K

using xlite as the softphone for vici.
Here is setting for account in xlite:


3. manual input call.
4. ill give shots on 2 output.
5. i don't know how to debug.
6. its not virtuals, installed on a machine solely.
7. sharing with other devices on a private network. portal only accessible within the local network at the moment.

[williamconley]

Xlite settings are not related unless your "2nd call agent" never gets sound even if he's the first call.

3. manual input call.

Describe your process of generating the call. Button for button.

You'll need to learn to use the asterisk CLI and especially SIP debugging. Google is your friend.