FOOD FOR THOUGHT Outbound Call Delay in Ringing Mobile Phone

[DesertEric]

Hello
I have searched the forums (possibly not using the right keywords) trying to find a similar issue. I have an external mobile (verizon) phone setup in one of my ring groups (5555551212#). When the specific option is pressed in the IVR the IP Phones ring immedietly, but I'd say the mobile phone rings only 10 to 20% of the time. I've verified with my DID/Trunk provider I don't have a lack of available channels and I've watched the asterisk cli so see if I can spot a difference when a call routes properly to the mobile and when it drops into space... and I don't see any difference.

Any advice? Other places to look?

Thanks!
Eric

 

[briankelly63]

The failure may be at a different level, like at the RTP level (firewall). Try some other numbers and see what happens. You may have to get more into the weeds. Do you know if the mobile rings and you can't hear the ringback or does the phone itself not ring.

 

[DesertEric]

Hi
Thanks for the reply. The phone itself does not ring. A firewall would cause intermittent issues like this. When I direct dial the mobile phone the string I see on the cli looks like the same sip/trunk/number... and direct dial always works

Thanks
Eric

 

[briankelly63]

So when you "direct dial" you are calling from a local pbx extension? What exactly do you dial when calling "direct"? When you use the IVR is that local too or is that from the outside? Need more detail.

 

[DesertEric]

Hello
Thanks for the RTP advise... I find that when the calls work, I get a "probation passed" message and additional lines of text made larger below... and when they fail I don't get the message. Only problem is that its totally random when it happens and does not happen.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/vitel-outbound/602xxxxxxx
-- SIP/vitel-outbound-00000043 is making progress passing it to Local/602xxxxxxx@from-internal-00000024;2
-- Local/602xxxxxxx@from-internal-00000024;1 is making progress passing it to Local/1003@from-queue-00000023;2
> 0xb76bd738 -- Probation passed - setting RTP source address to 64.2.xxx.xxx:xxxxx


When it fails it just looks like this

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/vitel-outbound/602xxxxxxx

This is my first PBX setup and I'm still very new to the whole thing, so my apologies in advance if I'm asking a stupid question or missing something obvious

Thanks
eric

 

[briankelly63]

May be a NAT issue or trunk config issue. Debugging can determine whats happening with the rtp streams but its complicated. Check in this forum or on google for"directmedia" or in older version "canreinvite" in the trunk config you have set up for vitelity