# Inbound / Outbound call ring issue **URL:** \ **Category:** Asterisk Support\ **Created:** [April 20, 2006, 1:14pm UTC](https://community.asterisk.org/t/inbound-outbound-call-ring-issue/5847 "2006-04-20T13:14:15Z")\ **Posts on this page:** 5\ **Page:** 1
**Author:** ![tanc08](https://avatars.discourse-cdn.com/v4/letter/t/b19c9b/32.png) [@tanc08](https://community.asterisk.org/u/tanc08)\ **Post date:** [April 20, 2006, 1:14pm UTC](https://community.asterisk.org/t/inbound-outbound-call-ring-issue/5847/1 "2006-04-20T13:14:15Z")
I have installed Asterisk@home 2.8 in CentOS with 2.8 Intel Pentium 4 and 512 RAM. I am using TDM400P with 1 FXO and 1 FXS. I connected POTS to FXO and an analog phone to FXS. I have setup that only my mobile phone can call my POTS, and vice versa. ## Trunk Configuration Outbound Caller ID: my POTS number with area code Maximum Channels: 1 Dial Rules: My mobile number Dial Rules Wizards: (pick one) Outbound Dial Prefix: None Zap Identifier: g0 ## Zap Extension Configuration Display Name: 1000 Outbound CID: None Emergency CID: None Record Incoming/Outgoing: On Demand Channel: 1 COntext: from-internal Signaling: fxo\_ks Echocancel: yes Echocancelwgenbridged: no Echotraining: 800 Busydetect: no Busycount: 7 Callprogress: no Dial: ZAP/1 Accountcode: None Mailbox: 1000@device voicemail password:12345 Email/Pager address: [1000@testing.com](mailto:1000@testing.com) Email attachment, Play CID, Play Envelope, Delete Vmail: yes Vm options: None Vm context: default ## Inbound Route Configuration DID Number, Caller ID Number, Fax Email: None Fax Extension: freePBX default Privacy Manager: No Immediate Answer: No Pause After Answer: 0 Alert Info: None Core: 1000 is selected for destination ## Outbound Route Configuration Route Name: PSTN Route Password: 12345 Emergency Dialing: None Dial Patterns: My mobile number Insert: Pick pre-defined patterns Trunk Sequence: ZAP/g0 I was comparing two method tests. ## 1st method test Without Asterisk and TDM400P involved. It was a regular analog phone called mobile, and vice versa. Mobile called analog phone - analog phone rang after first rang on mobile. Analog phone called mobile - analog phone rang after 5 seconds. ## 2nd method test With Asterisk and TDM400P card involved. POTS plugged to FXO. Analog phone plugged to FXS. Mobile called analog phone - analog phone rang between first and second rang on mobile. Analog phone called mobile - I dialed my mobile number, then IVR played to prompted password. I entered password with #. After 8 seconds, I heard rings. My question is how come 2nd method test, phone rang a little slower ? Is that a way to tweak it to be the same as in 1st method test ? passw ---
**Author:** ![RichardHH](https://avatars.discourse-cdn.com/v4/letter/r/a4c791/32.png) [@RichardHH](https://community.asterisk.org/u/RichardHH)\ **Post date:** [April 20, 2006, 1:43pm UTC](https://community.asterisk.org/t/inbound-outbound-call-ring-issue/5847/2 "2006-04-20T13:43:19Z")
Thats sounds like the typical digit/exten timeout asterisk is doing. I am not quite sure tho, where it sppears…since the IVR indicates (normally) that the dial was already initiated… Mobilenumber dialing closed with a “#” ? Any interessting CLI messages ? ---
**Author:** ![tanc08](https://avatars.discourse-cdn.com/v4/letter/t/b19c9b/32.png) [@tanc08](https://community.asterisk.org/u/tanc08)\ **Post date:** [April 20, 2006, 2:32pm UTC](https://community.asterisk.org/t/inbound-outbound-call-ring-issue/5847/3 "2006-04-20T14:32:03Z")
Richard, I don’t think # play a factor. I dialed the mobile number, then right away IVR prompted me to key in password. I am very new to Asterisk@Home. How do I see CLI message ? Any thing to turn on to see the debug ? ---
**Author:** ![RichardHH](https://avatars.discourse-cdn.com/v4/letter/r/a4c791/32.png) [@RichardHH](https://community.asterisk.org/u/RichardHH)\ **Post date:** [April 20, 2006, 2:38pm UTC](https://community.asterisk.org/t/inbound-outbound-call-ring-issue/5847/4 "2006-04-20T14:38:07Z")
open a terminal on the aah box and type asterisk -rvvvvv Then you have a CLI (command line interface) of asterisk with a verbose level of 5. ---
**Author:** ![RichardHH](https://avatars.discourse-cdn.com/v4/letter/r/a4c791/32.png) [@RichardHH](https://community.asterisk.org/u/RichardHH)\ **Post date:** [April 20, 2006, 2:40pm UTC](https://community.asterisk.org/t/inbound-outbound-call-ring-issue/5847/5 "2006-04-20T14:40:01Z")
PS.: Please post your real sip.conf,zapata.conf and extensions.conf. Your first posting is not refelcting the content of your conf files. “XXX” out passwords if needed 😄