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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: Graham on January 17, 2009, 03:45:07 PM

Title: Greetings
Post by: Graham on January 17, 2009, 03:45:07 PM

When trying to record a greeting using *60* the recording is in slow motion, but when playing back a previous greeting with  *61* it’s fine.

Title: Re: Greetings
Post by: Graham on January 17, 2009, 03:51:51 PM

Correction this also affects calls

Title: Re: Greetings
Post by: SARK devs on January 17, 2009, 03:55:14 PM

Are you running under VM?

Title: Re: Greetings
Post by: Graham on January 17, 2009, 04:25:19 PM

Nope not VM, it was working fine

I've started using a new phone Pirelli DP-L10

Title: Re: Greetings
Post by: Graham on January 17, 2009, 05:14:44 PM

Looked at this a bit more and it now seems the outbound sound works fine until I try and record a greeting then all calls after that the outbound sound slowed down.

Title: Re: Greetings
Post by: SARK devs on January 17, 2009, 06:25:58 PM

What release do you have?

Title: Re: Greetings
Post by: Graham on January 17, 2009, 06:29:55 PM

sail-2.2.1-708

Title: Re: Greetings
Post by: SARK devs on January 17, 2009, 08:34:31 PM

I can't recreate this Graham.  Works perfectly on our reference rig.  I'm not even sure where to go with it because this stuff is all raw asterisk.  Perhaps if you could relate what changed before the the problem emerged and also run a verbose console log of the recording process with agi debug turned on.  Maybe we'll be able to see some unexpected state change within asterisk.  Also, do you get the same results using a different phone? 

Best

S

Title: Re: Greetings
Post by: Graham on January 17, 2009, 09:20:37 PM

Only seems to do it with DP-L10, X-Lite seems fine.

I’ve tried a few things and this is what I ended up with.

1)   Received an incoming call, outbound sound was fine.
2)   Made call to *60* and recorded a greeting, playback was slowed down.
3)   Received an incoming call, outbound sound was slowed down same as with greeting.
4)   Turned DP-L10 off and on and everything works fine until you try and recorded a greeting.

When I say recorded a greeting it also does it when you dial another extn from the DP-L10 and leave a voicemail.

Below is the debug output, the first one is of me calling to create a greeting and the second one is showing the inbound call after making a call to create a greeting.

Greeting

server-1*CLI> agi debug
AGI Debugging Enabled
    -- Executing [*60*3003@internal:1] BackGround("SIP/5001-09375438", "silence/1") in new stack
    -- <SIP/5001-09375438> Playing 'silence/1' (language 'en')
    -- Executing [*60*3003@internal:2] AGI("SIP/5001-09375438", "selintra|*60*3003") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/5001-09375438
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1232222591.16
AGI Tx >> agi_callerid: 5001
AGI Tx >> agi_calleridname: Wireless
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: *60*3003
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: internal
AGI Tx >> agi_extension: *60*3003
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << EXEC Authenticate 0306
    -- AGI Script Executing Application: (Authenticate) Options: (0306)
    -- <SIP/5001-09375438> Playing 'agent-pass' (language 'en')
    -- <SIP/5001-09375438> Playing 'auth-thankyou' (language 'en')
AGI Tx >> 200 result=0
AGI Rx << EXEC Playback pm-announcement-number
    -- AGI Script Executing Application: (Playback) Options: (pm-announcement-number)
[Jan 17 20:03:21] WARNING[9247]: file.c:602 ast_openstream_full: File pm-announcement-number does not exist in any format
[Jan 17 20:03:21] WARNING[9247]: file.c:912 ast_streamfile: Unable to open pm-announcement-number (format 0x4 (ulaw)): No such file or directory
[Jan 17 20:03:21] WARNING[9247]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/5001-09375438 for pm-announcement-number
AGI Tx >> 200 result=0
AGI Rx << EXEC SayDigits 3003
    -- AGI Script Executing Application: (SayDigits) Options: (3003)
    -- <SIP/5001-09375438> Playing 'digits/3' (language 'en')
    -- <SIP/5001-09375438> Playing 'digits/0' (language 'en')
    -- <SIP/5001-09375438> Playing 'digits/0' (language 'en')
    -- <SIP/5001-09375438> Playing 'digits/3' (language 'en')
AGI Tx >> 200 result=0
AGI Rx << EXEC Playback is-now-being-recorded
    -- AGI Script Executing Application: (Playback) Options: (is-now-being-recorded)
[Jan 17 20:03:23] WARNING[9247]: file.c:602 ast_openstream_full: File is-now-being-recorded does not exist in any format
[Jan 17 20:03:23] WARNING[9247]: file.c:912 ast_streamfile: Unable to open is-now-being-recorded (format 0x4 (ulaw)): No such file or directory
[Jan 17 20:03:23] WARNING[9247]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/5001-09375438 for is-now-being-recorded
AGI Tx >> 200 result=0
AGI Rx << EXEC Playback press-pound-save-changes
    -- AGI Script Executing Application: (Playback) Options: (press-pound-save-changes)
[Jan 17 20:03:23] WARNING[9247]: file.c:602 ast_openstream_full: File press-pound-save-changes does not exist in any format
[Jan 17 20:03:23] WARNING[9247]: file.c:912 ast_streamfile: Unable to open press-pound-save-changes (format 0x4 (ulaw)): No such file or directory
[Jan 17 20:03:23] WARNING[9247]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/5001-09375438 for press-pound-save-changes
AGI Tx >> 200 result=0
AGI Rx << RECORD FILE usergreetingtemp gsm "#" 30000 BEEP s=10
    -- <SIP/5001-09375438> Playing 'beep' (language 'en')
AGI Tx >> 200 result=35 (dtmf) endpos=220320
AGI Rx << STREAM FILE usergreetingtemp "" 0
    -- Playing 'usergreetingtemp' (escape_digits=) (sample_offset 0)
AGI Tx >> 200 result=0 endpos=220320
AGI Rx << STREAM FILE save-announce-press "123" 0
[Jan 17 20:04:01] WARNING[9247]: file.c:602 ast_openstream_full: File save-announce-press does not exist in any format
AGI Tx >> 200 result=0 endpos=0
AGI Rx << STREAM FILE digits/1 "123" 0
    -- Playing 'digits/1' (escape_digits=123) (sample_offset 0)
AGI Tx >> 200 result=0 endpos=5280
AGI Rx << STREAM FILE to-rerecord-announce "123" 0
[Jan 17 20:04:01] WARNING[9247]: file.c:602 ast_openstream_full: File to-rerecord-announce does not exist in any format
AGI Tx >> 200 result=0 endpos=0
AGI Rx << STREAM FILE digits/2 "123" 0
    -- Playing 'digits/2' (escape_digits=123) (sample_offset 0)
AGI Tx >> 200 result=0 endpos=4800
AGI Rx << STREAM FILE to-cancel-this-msg "123" 0
[Jan 17 20:04:02] WARNING[9247]: file.c:602 ast_openstream_full: File to-cancel-this-msg does not exist in any format
AGI Tx >> 200 result=0 endpos=0
AGI Rx << STREAM FILE press "123" 0
[Jan 17 20:04:02] WARNING[9247]: file.c:602 ast_openstream_full: File press does not exist in any format
AGI Tx >> 200 result=0 endpos=0
AGI Rx << STREAM FILE digits/3 "123" 0
    -- Playing 'digits/3' (escape_digits=123) (sample_offset 0)
AGI Tx >> 200 result=0 endpos=4800
AGI Rx << STREAM FILE silence/5 "123" 0
    -- Playing 'silence/5' (escape_digits=123) (sample_offset 0)
AGI Tx >> 200 result=49 endpos=14080
AGI Rx << EXEC Playback your-msg-has-been-saved
    -- AGI Script Executing Application: (Playback) Options: (your-msg-has-been-saved)
[Jan 17 20:04:04] WARNING[9247]: file.c:602 ast_openstream_full: File your-msg-has-been-saved does not exist in any format
[Jan 17 20:04:04] WARNING[9247]: file.c:912 ast_streamfile: Unable to open your-msg-has-been-saved (format 0x4 (ulaw)): No such file or directory
[Jan 17 20:04:04] WARNING[9247]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/5001-09375438 for your-msg-has-been-saved
AGI Tx >> 200 result=0
AGI Rx << EXEC Playback goodbye
    -- AGI Script Executing Application: (Playback) Options: (goodbye)
[Jan 17 20:04:04] WARNING[9247]: file.c:602 ast_openstream_full: File goodbye does not exist in any format
[Jan 17 20:04:04] WARNING[9247]: file.c:912 ast_streamfile: Unable to open goodbye (format 0x4 (ulaw)): No such file or directory
[Jan 17 20:04:04] WARNING[9247]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/5001-09375438 for goodbye
AGI Tx >> 200 result=0
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/5001-09375438' status is 'UNKNOWN'
    -- Executing [h@internal:1] Hangup("SIP/5001-09375438", "") in new stack
  == Spawn extension (internal, h, 1) exited non-zero on 'SIP/5001-09375438'
agi no debug
AGI Debugging Disabled

Incoming Call

server-1*CLI> agi debug
AGI Debugging Enabled
    -- Executing [9883008@mainmenu:1] AGI("SIP/9883008-09375438", "selintra|Inbound|9883008") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/9883008-09375438
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1232223165.17
AGI Tx >> agi_callerid: xxxxxxxxxxx
AGI Tx >> agi_calleridname: xxxxxxxxxxx
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 9883008
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: mainmenu
AGI Tx >> agi_extension: 9883008
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> LI>
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE RINGDELAY
AGI Tx >> 200 result=1 (0)
AGI Rx << GET VARIABLE FAXDETECT
AGI Tx >> 200 result=1 (2)
AGI Rx << GET VARIABLE CALLRECORD2
AGI Tx >> 200 result=1 (None)
AGI Rx << GET VARIABLE LTERM
AGI Tx >> 200 result=1 (NO)
AGI Rx << SET VARIABLE MOH ""
AGI Tx >> 200 result=1
AGI Rx << SET CALLERID xxxxxxxxxxx
AGI Tx >> 200 result=1
AGI Rx << DATABASE GET "STAT" "OCSTAT"
AGI Tx >> 200 result=0
AGI Rx << SET VARIABLE REMOTENUM "9883008"
AGI Tx >> 200 result=1
AGI Rx << GET VARIABLE MTIME
AGI Tx >> 200 result=1 (ON)
AGI Rx << GET VARIABLE IFTIME(22:00-09:00|*|*|*?CLOSED:OPEN)
AGI Tx >> 200 result=1 (OPEN)
AGI Rx << GET VARIABLE VOICEINSTR
AGI Tx >> 200 result=1 (YES)
AGI Rx << SET VARIABLE OPEN "YES"
AGI Tx >> 200 result=1
AGI Rx << DATABASE GET "STAT" "IVRSTAT"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE SYSOP
AGI Tx >> 200 result=1 (1001)
AGI Rx << SET PRIORITY 1
AGI Tx >> 200 result=0
AGI Rx << SET EXTENSION 5001
AGI Tx >> 200 result=0
AGI Rx << SET CONTEXT internal
AGI Tx >> 200 result=0
    -- AGI Script selintra completed, returning 0
    -- Executing [5001@internal:1] AGI("SIP/9883008-09375438", "selintra|OutCluster|5001") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/9883008-09375438
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1232223165.17
AGI Tx >> agi_callerid: xxxxxxxxxxx
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 9883008
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: internal
AGI Tx >> agi_extension: 5001
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << SET PRIORITY 1
AGI Tx >> 200 result=0
AGI Rx << SET EXTENSION 5001
AGI Tx >> 200 result=0
AGI Rx << SET CONTEXT default
AGI Tx >> 200 result=0
    -- AGI Script selintra completed, returning 0
    -- Executing [5001@default:1] AGI("SIP/9883008-09375438", "selintra|InCall|") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
AGI Tx >> agi_request: selintra
AGI Tx >> agi_channel: SIP/9883008-09375438
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1232223165.17
AGI Tx >> agi_callerid: xxxxxxxxxxx
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 9883008
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: default
AGI Tx >> agi_extension: 5001
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << GET VARIABLE BLINDTRANSFER
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE VOICEINSTR
AGI Tx >> 200 result=1 (YES)
AGI Rx << DATABASE GET "STAT" "OCSTAT"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE MTIME
AGI Tx >> 200 result=1 (ON)
AGI Rx << DATABASE GET "cfimopen" "5001"
AGI Tx >> 200 result=0
AGI Rx << DATABASE GET "cfim" "5001"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << DATABASE GET "STAT" "OCSTAT"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE MTIME
AGI Tx >> 200 result=1 (ON)
AGI Rx << GET VARIABLE VOICEINSTR
AGI Tx >> 200 result=1 (YES)
AGI Rx << DATABASE GET "cfimopen" "5001"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << DATABASE GET "STAT" "OCSTAT"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE MTIME
AGI Tx >> 200 result=1 (ON)
AGI Rx << GET VARIABLE VOICEINSTR
AGI Tx >> 200 result=1 (YES)
AGI Rx << DATABASE GET "cfim" "5001"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE EXTLEN
AGI Tx >> 200 result=1 ()
AGI Rx << DATABASE GET "ringdelay" "5001"
AGI Tx >> 200 result=0
AGI Rx << GET VARIABLE INTRINGDELAY
AGI Tx >> 200 result=1 (30)
AGI Rx << GET VARIABLE CALLRECORD1
AGI Tx >> 200 result=1 (None)
AGI Rx << GET VARIABLE CALLRECORD2
AGI Tx >> 200 result=1 (None)
AGI Rx << GET VARIABLE MOH
AGI Tx >> 200 result=1 ()
AGI Rx << EXEC Dial SIP/5001|30|t
    -- AGI Script Executing Application: (Dial) Options: (SIP/5001|30|t)
    -- Called 5001
    -- SIP/5001-093a5c10 is ringing
    -- SIP/5001-093a5c10 answered SIP/9883008-09375438
AGI Tx >> 200 result=-1
  == Spawn extension (default, 5001, 1) exited non-zero on 'SIP/9883008-09375438'
    -- Executing [h@default:1] Hangup("SIP/9883008-09375438", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/9883008-09375438'
agi no debug
AGI Debugging Disabled

Title: Re: Greetings
Post by: SARK devs on January 17, 2009, 10:04:10 PM

ok - nothing obvious there.  Usually if a digital recording sounds slow it's caused by having the record sample rate set higher than the play back algorithm is expecting.  Some of the newer phones on the market support HD sound (for example the new SNOMs have it).  I'm wondering if maybe your DP-L10 has such a feature and you are perhaps inadvertently turning it on and off with our feature codes?


S

Title: Re: Greetings
Post by: Graham on January 17, 2009, 10:21:00 PM

Is this a phone bug, tried an echo test and it's fine.

This issue only seems to happen when making a recording.

How do you reset asterisk stop start button doesn’t seem to be working.

Title: Re: Greetings
Post by: SARK devs on January 17, 2009, 10:46:32 PM

I don't know if it is a phone bug but I'm pretty sure it is a phone issue.  STOP/START won#t always work if you've been using the asterisk console.

To do a clean restart; at the Linux console do...

Code: [Select]

/etc/init.d/sark stop
/etc/init.d/sark start
Best

S

Title: Re: Greetings
Post by: Graham on January 17, 2009, 11:15:49 PM

Seems restarting asterisk doesn’t fix the outbound call speed only restarting the phone does.

Also If I leave a voicemail and at the end don’t press # then the next inbound calls outbound sound is fine, but if I do press # at the end of a voicemail then it isn’t

Even more odd if I press # during a call it does a transfer that’s fine, and I abort the transfer but after that the outbound sound is broken, so it’s something to do with pressing #

Also I get this when starting asterisk:

[root@server-1 ~]# /etc/init.d/sark start
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
FATAL: Module zaptel not found.
Waiting for udev.
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

Starting asterisk:                                         [  OK  ]

Title: Re: Greetings
Post by: SARK devs on January 17, 2009, 11:44:18 PM

Code: [Select]

[root@server-1 ~]# /etc/init.d/sark start
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature

you can ignore the above messages it has something to do with the way in which ATrpms build their rpms - its annoying but benign.

Code: [Select]

FATAL: Module zaptel not found.
Waiting for udev.
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected
This is more worrying, you either do not have zaptel installed or you have done a yum update and downloaded a new kernel without handling zaptel-kmdl.

what does 'uname -r' return at the linux console?

what does 'rpm-qa | grep zaptel-kmdl' return?

Neither of these things should mess up playback speed (although they may make it a little choppy), however its a good idea to fix it before you do anything else.

You can turn off hash processing (at least for transfers)in headers->features.conf...  set blindxfer=##

Did you check the phone manual to see what # does (if anything) and whether it supports highdef audio?

Best

S

 

 

Title: Re: Greetings
Post by: Graham on January 17, 2009, 11:57:10 PM

All the updates I do are via SME

[root@server-1 ~]# uname -r
2.6.9-78.0.8.EL

[root@server-1 ~]# rpm -qa | grep zaptel-kmdl
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature

Title: Re: Greetings
Post by: Teviot on January 18, 2009, 12:17:04 AM

If I'm not mistake and Selintra will confirm or deny this you should have got something like this

Code: [Select]

[root@sail ~]# uname -r
2.6.9-78.0.8.EL
[root@sail ~]# rpm -qa | grep zaptel-kmdl
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
zaptel-kmdl-2.6.9-67.0.1.EL-1.4.9.2-48.el4
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
zaptel-kmdl-2.6.9-67.0.1.ELsmp-1.4.9.2-48.el4
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
[root@sail ~]#
And ignoring the signature errors

Title: Re: Greetings
Post by: SARK devs on January 18, 2009, 12:35:02 AM

Teviot is correct

You don't appear to have installed zaptel or zaptel-kmdl.  I also notice fom earlier that you don't have the extended sound pack installed either.

Kind Regards

S

Title: Re: Greetings
Post by: Graham on January 18, 2009, 12:41:41 AM

Think I've fixed it must have mised it when I installed it, yum install zaptel-kmdl-`uname -r` --enablerepo=atrpms

I've just run these again, and the only one that did anything was the one above all the other ones reported nothing to do

yum install zaptel --enablerepo=atrpms
yum install libpri  --enablerepo=atrpms
yum install asterisk --enablerepo=atrpms
yum install zaptel-kmdl-`uname -r` --enablerepo=atrpms
yum install asterisk-addons  --enablerepo=atrpms

does that mean the extended sound pack is installed

Title: Re: Greetings
Post by: Teviot on January 18, 2009, 12:44:56 AM

sprattgraham

Issue the following at the command prompt and check to see if you get a similar result as I did

uname -r

rpm -qa | grep zaptel-kmdl

Title: Re: Greetings
Post by: Graham on January 18, 2009, 12:55:51 AM

I can only see one difference, zaptel-kmdl-2.6.9-67.0.1.ELsmp-1.4.9.2-48.el4

[root@server-1 ~]# uname -r
2.6.9-78.0.8.EL

[root@server-1 ~]# rpm -qa | grep zaptel-kmdl
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
zaptel-kmdl-2.6.9-78.0.8.EL-1.4.12.1-54.99.el4
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature

Title: Re: Greetings
Post by: Teviot on January 18, 2009, 01:09:20 AM

Sprattgraham

Maybe run these again

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yum install zaptel --enablerepo=atrpms
yum install libpri  --enablerepo=atrpms
yum install asterisk --enablerepo=atrpms
yum install zaptel-kmdl-`uname -r` --enablerepo=atrpms
yum install asterisk-addons  --enablerepo=atrpms
then run

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uname -r and

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rpm -qa | grep zaptel-kmdland see what you get.

Other then that I don't have enought knowledge about SAIL to advise you any further than I have.  Selintra or someone with a little knowledge in this area would be the ones to help you with this.

Good Luck

Title: Re: Greetings
Post by: SARK devs on January 18, 2009, 01:20:25 AM

UK GB soundpack is here...

http://www.sarkpbx.com/sail/languagepack/sme-ast-en-uk-gpl-sounds-1.0.0-3.noarch.rpm

download it with wget and install it with rpm...

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wget http://www.sarkpbx.com/sail/languagepack/sme-ast-en-uk-gpl-sounds-1.0.0-3.noarch.rpm
rpm -Uvh sme-ast-en-uk-gpl-sounds-1.0.0-3.noarch.rpm
Regards

Title: Re: Greetings
Post by: Graham on January 18, 2009, 01:26:22 AM

Thanks again that sound pack is a lot better, do I need zaptel-kmdl-2.6.9-67.0.1.ELsmp-1.4.9.2-48.el4 or just zaptel-kmdl-2.6.9-67.0.1.EL-1.4.9.2-48.el4

I’ve checked the manual for my phone and it doesn’t say anything about HD, http://www.pirellibroadband.com/web/products-solutions/solutions/sme-net/terminals/default.page

Title: Re: Greetings
Post by: SARK devs on January 18, 2009, 10:59:16 AM

The two kmdl versions; EL and ELsmp are for the two linux kernels EL (single processor) and ELsmp (Symmetrical Multi Processor).  As a general rule, run the smp kernel if you have a multicore CPU(s) and the EL kernel if you have a single CPU.   If you don't know what you've got then look in /proc/cpuinfo - like this....

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cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 15
model           : 2
model name      : Intel(R) Pentium(R) 4 CPU 2.80GHz
stepping        : 9
cpu MHz         : 2802.630
cache size      : 512 KB
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 2
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr
bogomips        : 5603.45
 

This will give you a summary of each CPU in the box.  The above example shows a single 2.8Ghz Pentium IV.

Title: Re: Greetings
Post by: Graham on January 18, 2009, 02:09:19 PM

Had a play with this # issue I changed all the SIP settings to sipgate and made a call and had no issue if pressing the # key, they changed them back to Asterisk without restarting the phone and the outbound audio was no good after restarting it was fine again.

Title: Re: Greetings
Post by: Graham on January 19, 2009, 03:15:24 AM

After a bit more of a play it now seems that pressing any key causes this outbound issue, but only when using Asterisk.

Now from reading this post, this user isn't having any issues: http://forums.digium.com/viewtopic.php?t=14924&highlight=dpl10

Title: Re: Greetings
Post by: Graham on January 24, 2009, 09:24:08 PM

Anyone have any ideas

Title: Re: Greetings
Post by: SARK devs on January 24, 2009, 11:15:30 PM

If it's only doing it on one phone type then it's phone related.  I can't recreate any of the symptoms you recount on our reference server and we've had no other reports of this nature so I'm satisfied that it definitely isn't SAIL related and it probably isn't Asterisk related. 

As an aside, it is generally NOT good practice to use # and *2 to do blind and attended transfers in the Asterisk environment, unless you have no choice.   Both of these depend upon the remote asterisk recognising the DTMF you are sending (which it may not always do).  Better to use SIP re-invites.  Pretty much ALL SIP phones (with the exception of one or two handheld DECT hybrids) are capable of SIP re-invite.  Usually this is done through a transfer button on the phone itself.  It involves no DTMF and, because the phone is in direct control, its usually much easier to retrieve an unanswered call or to abandon an incorrectly keyed transfer.   It is also less prone to losing calls "somewhere in the ether", which can be done quite easily with # and 2* transfers.

Kind Regards

S

Title: Re: Greetings
Post by: Graham on January 24, 2009, 11:20:35 PM

Thanks, I don't think it's the phone as it doesn't cause issues with sipgate.

Maybe an Asterisk issue, it's not just # or * it's the number buttons as well that cause the issue.

Title: Re: Greetings
Post by: Graham on February 05, 2009, 12:11:20 AM

Sorry to keep going on about this, is anyone know of an Asterisk server I can use to do an echo test to see if it’s my server.

Title: Re: Greetings
Post by: Teviot on February 05, 2009, 12:45:55 AM

Most of the sip & iax providers have a number you can call that does exactly that.

Search or call your provider(s) and ask them for the number to their echo test

Title: Re: Greetings
Post by: Graham on February 05, 2009, 12:51:47 AM

Need someone who is using Asterisk, tried with Sipgate and don't have any issues.