Call Menu Concern

[benghoi]

Hi

does somebody encounter this problem:

[Jan 20 00:30:26] WARNING[28442]: format_wav.c:156 check_header: Unexpected frequency 44100
[Jan 20 00:30:26] WARNING[28442]: file.c:386 fn_wrapper: Unable to open format wav
[Jan 20 00:30:26] WARNING[28442]: file.c:995 ast_streamfile: Unable to open intro (format 0x100 (g729)): No such file or directory
[Jan 20 00:30:26] WARNING[28442]: pbx.c:5830 pbx_builtin_background: ast_streamfile failed on SIP/AccelaSansay1-0000023b for intro

Please Help...

[bmorrison]

You have not converted your sound file into a format that asterisk can read.

If you're using a mac or linux operating system, it's easy to convert your file using sox (assuming you've installed the sox package):

sox -V foo.wav -r 8000 -c 1 -t ul -2 foo.ulaw

[benghoi]

The sound file is already working, thanks to bmorrison. But there is one thing, i can hear now the intro voice prompt but when im trying to select an option, there is nothing, it won't route to option 1, it only play the intro voice again.

[benghoi]

This is my setup on Call Menu:

MenuID: Main
Menu Name: Welcome Message
Menu Prompt: intro.wav
MEnu Time: 10
Menu Timeout: none
Menu Invalid Prompt: none
Menu Repeat: 2
Menu Time Check: 0-No Time Check
Call Time: 24hours
Track Calls in: 1-Realtime Tracking
Tracking Group: CALLMENU
Log Key Press: 0-No DTMF Logging

Call Menu Option:

Option 0: Route: HANGUP
Option 1: Route: PHONE Phone: 1000
Option 2: Route: PHONE Phone: 1001

[benghoi]

i already tried what's in the manual but still no output. i can reach to intro prompt but i can't select any option, the intro prompt will play over and over again... please help...

[benghoi]

This is the CLI logs when im trying to call the number:

[Jan 28 02:11:07] -- Called 18888178593@AccelaSansay1
[Jan 28 02:11:24] -- Executing [8888178593@trunkinbound:1] AGI("SIP/AccelaSansay1-00000002", "agi-DID_route.agi") in new stack
[Jan 28 02:11:24] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Jan 28 02:11:24] -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20120128021124_8888178593_8888178593)
[Jan 28 02:11:24] ERROR[3169]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Jan 28 02:11:24] ERROR[3169]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Jan 28 02:11:24] -- AGI Script agi-DID_route.agi completed, returning 0
[Jan 28 02:11:24] -- Executing [s@Main:1] AGI("SIP/AccelaSansay1-00000002", "agi-VDAD_inbound_calltime_check.agi|CALLMENU-----YES-----Main--------------------") in new stack
[Jan 28 02:11:24] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
[Jan 28 02:11:24] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 28 02:11:24] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jan 28 02:11:24] -- AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
[Jan 28 02:11:24] -- Executing [s@Main:2] Set("SIP/AccelaSansay1-00000002", "INVCOUNT=0") in new stack
[Jan 28 02:11:24] -- Executing [s@Main:3] BackGround("SIP/AccelaSansay1-00000002", "sample") in new stack
[Jan 28 02:11:24] -- <SIP/AccelaSansay1-00000002> Playing 'sample' (language 'en')
[Jan 28 02:11:26] -- SIP/AccelaSansay1-00000001 is making progress passing it to SIP/1000-00000000
[Jan 28 02:11:26] -- SIP/AccelaSansay1-00000001 answered SIP/1000-00000000
[Jan 28 02:11:26] -- Packet2Packet bridging SIP/1000-00000000 and SIP/AccelaSansay1-00000001
[Jan 28 02:11:42] -- Executing [s@Main:4] BackGround("SIP/AccelaSansay1-00000002", "for-sales") in new stack
[Jan 28 02:11:42] -- <SIP/AccelaSansay1-00000002> Playing 'for-sales' (language 'en')
[Jan 28 02:11:43] -- Executing [s@Main:5] BackGround("SIP/AccelaSansay1-00000002", "press") in new stack
[Jan 28 02:11:43] -- <SIP/AccelaSansay1-00000002> Playing 'press' (language 'en')
[Jan 28 02:11:43] -- Executing [s@Main:6] BackGround("SIP/AccelaSansay1-00000002", "./digits/1") in new stack
[Jan 28 02:11:43] -- <SIP/AccelaSansay1-00000002> Playing './digits/1' (language 'en')
[Jan 28 02:11:44] -- Executing [s@Main:7] BackGround("SIP/AccelaSansay1-00000002", "for-tech-support") in new stack
[Jan 28 02:11:44] -- <SIP/AccelaSansay1-00000002> Playing 'for-tech-support' (language 'en')
[Jan 28 02:11:45] -- Executing [s@Main:8] BackGround("SIP/AccelaSansay1-00000002", "press") in new stack
[Jan 28 02:11:45] -- <SIP/AccelaSansay1-00000002> Playing 'press' (language 'en')
[Jan 28 02:11:46] -- Executing [s@Main:9] BackGround("SIP/AccelaSansay1-00000002", "./digits/2") in new stack
[Jan 28 02:11:46] -- <SIP/AccelaSansay1-00000002> Playing './digits/2' (language 'en')
[Jan 28 02:11:46] -- Executing [s@Main:10] BackGround("SIP/AccelaSansay1-00000002", "for-hungup") in new stack
[Jan 28 02:11:46] WARNING[3169]: file.c:665 ast_openstream_full: File for-hungup does not exist in any format
[Jan 28 02:11:46] WARNING[3169]: file.c:995 ast_streamfile: Unable to open for-hungup (format 0x100 (g729)): No such file or directory
[Jan 28 02:11:46] WARNING[3169]: pbx.c:5830 pbx_builtin_background: ast_streamfile failed on SIP/AccelaSansay1-00000002 for for-hungup
[Jan 28 02:11:46] -- Executing [s@Main:11] BackGround("SIP/AccelaSansay1-00000002", "press") in new stack
[Jan 28 02:11:46] -- <SIP/AccelaSansay1-00000002> Playing 'press' (language 'en')
[Jan 28 02:11:47] -- Executing [s@Main:12] BackGround("SIP/AccelaSansay1-00000002", "./digits/0") in new stack
[Jan 28 02:11:47] -- <SIP/AccelaSansay1-00000002> Playing './digits/0' (language 'en')
[Jan 28 02:11:48] -- Executing [s@Main:13] WaitExten("SIP/AccelaSansay1-00000002", "5") in new stack
[Jan 28 02:11:53] -- Timeout on SIP/AccelaSansay1-00000002, continuing...
[Jan 28 02:11:53] -- Executing [s@Main:14] BackGround("SIP/AccelaSansay1-00000002", "sample") in new stack
[Jan 28 02:11:53] -- <SIP/AccelaSansay1-00000002> Playing 'sample' (language 'en')
[Jan 28 02:11:57] -- Executing [h@default:1] DeadAGI("SIP/1000-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----50-----31") in new stack
[Jan 28 02:11:57] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -50-----31 completed, returning 0
[Jan 28 02:11:57] == Spawn extension (default, 18888178593, 1) exited non-zero on 'SIP/1000-00000000'
[Jan 28 02:11:59] == Spawn extension (Main, s, 14) exited non-zero on 'SIP/AccelaSansay1-00000002'
[Jan 28 02:11:59] -- Executing [h@Main:1] DeadAGI("SIP/AccelaSansay1-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jan 28 02:11:59] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[Op3r]

DTMF and g729.

Use g711 if you are doing DTMF. g729 has issues with dtmf.

[williamconley]

check your sip settings for the trunk and verify with your provider which dtmfmode you should be using. of course, there are only a few so you COULD just try them all until you get lucky.

[benghoi]

they are using dtmfmode=rfc2833

type=friend
host=69.94.226.174
canreinvite=no
disallow=all
allow=g729
allow=ulaw
dtmfmode=rfc2833

i have no idea about g711, can you give me some details about g711.

[williamconley]

g711 is ulaw

[benghoi]

its working now but there is still one thing that is not working. hehehe...

Guys,
can anybody knows how to setup direct dial extension number in call menu, sample call menu prompt:

Welcome to our company, if you know the extension of the person you calling please dial it now

press 1 for department 1
press 2 for department 2
press 0 for hungup

question:
how to setup the underline phrase?
is phone directory available to vicidial?

[yuvrajkc]

You might want to try using FreePBX with vicidial for a better IVR options.

[mflorell]

Setting that up is detailed in the Vicidial manager manual