[Spce-user] Voicemail - '*' interrupt not working and message not playing from phone
Tomi Hakkarainen
tomishakkarainen at gmail.com
Thu Jan 10 17:08:36 EST 2019
Hi,
Default configuration the call is routed to different asterisk application when call is routed/forwarded to subscriber voicemail …
It could be changed to the same as the internal calls are routed when you call from subscriber extension…
Change the correct context parameter on asterisk extensions.config
VoiceMail -> VoicemailMain
Maybe you have custom config on /etc/ngcp-config/templates/etc/asterisk/ on that old “working” setup which you could check for hints?
For the correct solution you should create custom file on that folder and adjust it to fit your needs.
I tested simply adjusting the /etc/asterisk/extension.conf under for example [voicemailcaller_unavail] context
; Play default
;exten => s,n(default),VoiceMail(${ARG1},us)
exten => s,n(default),VoiceMailMain(s${ARG1},us)
and it started to ask password after I pressed ‘*’ for the call which was forwarded to subscribers voicemail when extension is not registered…
I hope this makes some sense and you can work out and figure the proper solution you could use on your setup…
Tomi
> On 10 Jan 2019, at 16.52, Joe Gaylord <joe at torchlake.com> wrote:
>
> Yes, when doing a packet capture and analyzing it in wireshark, I am seeing an RTP Event with ID of DTMF Star *. Tried multiple endpoints on the PSTN with same result. All 3 sipwise platforms use the same provider (the one working one and the two non-working setups). Seems also, when I hit the # key Asterisk is recognizing this keypress on the PSTN side, just not the Star key.
>
> I welcome any other ideas.
>
> Thank You,
> Joe Gaylord
> COLI Inc.
>
>
>
> On 1/9/2019 3:25 PM, Tomi Hakkarainen wrote:
>> Hi,
>>
>> Have you checked that you get the ’*’ from the PSTN to the new setup?
>> I mean by capturing the network traffic and analyzing the results.
>>
>> If using SIP INFO to deliver dtmf you should see SIP INFO messages,
>> If telephone-events rfc2833 there should be udp packets with in the media stream only with different payloadtype.
>>
>> At least wireshark is capable to identify those all so I would suggest to try with that.
>>
>> If inband you would have to listen the RTP to hear ’beep’
>>
>>
>> Also have you tried to call with different endpoints on PSTN do all give the same results?
>>
>> I assume you have the same service provider and connection to PSTN on the new setup and the old ones...
>>
>>
>> Tomi
>>
>> On 9 Jan 2019, at 21.38, Joe Gaylord <joe at torchlake.com> wrote:
>>
>> So if the sound files are installed and they are playing fine when dialing ext 2000 from a internally registered extension, does this mean the sound files are correct? I guess I am having a hard time determining why sound files missing would cause the user not to be able to access voicemail externally by pressing the * (Asterisk) key. When the * (Asterisk) key is pressed during the 'unavailable greeting' with an outside PSTN call, nothing happens. From the manual it shows after the asterisks key is pressed, the system should ask for the users PIN to gain access to the voicemail system. When viewing the asterisk -vvvvvr console, I am not getting any error or any acknowledgement of any '*' key presses.
>>
>> This feature is listed in the Manual Docs, which is where I am having the problem. This feature works in my oldest SPCE install (~mr3.2.1.2), but not the last two installs I have done.
>> "
>> • The user can also dial his own number from PSTN, if he setup Call Forward on Not Available to the Voicebox, and when reaching the voicemail server he can interrupt the "user is unavailable" message by pressing * key and then be prompted for the PIN. After entering PIN and confirming with # key he will enter own voicemail menu. PIN is random by default and must be kept secret for that reason
>> "
>> asterisk -vvvvvr
>> Asterisk 13.14.1~dfsg-2+deb9u4, Copyright (C) 1999 - 2014, Digium, Inc. and others.
>> Created by Mark Spencer <markster at digium.com>
>> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
>> This is free software, with components licensed under the GNU General Public
>> License version 2 and other licenses; you are welcome to redistribute it under
>> certain conditions. Type 'core show license' for details.
>> =========================================================================
>> Output of the Asterisk console with asterisk -vvvvvr
>> "Connected to Asterisk 13.14.1~dfsg-2+deb9u4 currently running on spce (pid = 24516)
>> == Using SIP RTP CoS mark 5
>> > 0x7f56e4005120 -- Strict RTP learning after remote address set to: 216.245.XXX.YYY:32038
>> -- Executing [vmu12315184073 at sip_in:1] NoOp("SIP/sip_proxy-00000000", "") in new stack
>> -- Executing [vmu12315184073 at sip_in:2] Set("SIP/sip_proxy-00000000", "__LANG=en") in new stack
>> -- Executing [vmu12315184073 at sip_in:3] Set("SIP/sip_proxy-00000000", "__UUID=1166d839-8af4-4b99-8890-2ef7fdb07d16") in new stack
>> -- Executing [vmu12315184073 at sip_in:4] Set("SIP/sip_proxy-00000000", "CHANNEL(language)=en") in new stack
>> -- Executing [vmu12315184073 at sip_in:5] Gosub("SIP/sip_proxy-00000000", "voicemailcaller_unavail,s,1(1166d839-8af4-4b99-8890-2ef7fdb07d16)") in new stack
>> -- Executing [s at voicemailcaller_unavail:1] Answer("SIP/sip_proxy-00000000", "") in new stack
>> -- Executing [s at voicemailcaller_unavail:2] Set("SIP/sip_proxy-00000000", "TIMEOUT(digit)=5") in new stack
>> -- Digit timeout set to 5.000
>> -- Executing [s at voicemailcaller_unavail:3] Set("SIP/sip_proxy-00000000", "TIMEOUT(response)=10") in new stack
>> -- Response timeout set to 10.000
>> -- Executing [s at voicemailcaller_unavail:4] Wait("SIP/sip_proxy-00000000", "1") in new stack
>> -- Executing [s at voicemailcaller_unavail:5] VoiceMail("SIP/sip_proxy-00000000", "1166d839-8af4-4b99-8890-2ef7fdb07d16,us") in new stack
>> [Jan 4 14:22:12] NOTICE[24758][C-00000000]: apps/app_voicemail.c:4209 find_user_realtime_by_alias: Found mailbox '12315184073' for alias '1166d839-8af4-4b99-8890-2ef7fdb07d16'
>> -- <SIP/sip_proxy-00000000> Playing '/var/spool/asterisk/voicemail/default/1166d839-8af4-4b99-8890-2ef7fdb07d16/unavail.gsm' (language 'en')
>> > 0x7f56e4005120 -- Strict RTP switching to RTP target address 216.245.XXX.YYY:32038 as source
>> > 0x7f56e4005120 -- Strict RTP learning complete - Locking on source address 216.245.XXX.YYY:32038
>> -- <SIP/sip_proxy-00000000> Playing 'beep.slin' (language 'en')
>> -- Recording the message
>> -- x=0, open writing: /var/spool/asterisk/voicemail/default/1166d839-8af4-4b99-8890-2ef7fdb07d16/tmp/7f126m format: wav49, 0x7f56f0003340
>> -- x=1, open writing: /var/spool/asterisk/voicemail/default/1166d839-8af4-4b99-8890-2ef7fdb07d16/tmp/7f126m format: wav, 0x7f56f00052f0
>> -- User ended message by pressing #
>> -- <SIP/sip_proxy-00000000> Playing 'auth-thankyou.slin' (language 'en')
>> == Parsing '/var/spool/asterisk/voicemail/default/1166d839-8af4-4b99-8890-2ef7fdb07d16/INBOX/msg0014.txt': Found
>> -- Executing [s at voicemailcaller_unavail:6] Hangup("SIP/sip_proxy-00000000", "") in new stack
>> == Spawn extension (voicemailcaller_unavail, s, 6) exited non-zero on 'SIP/sip_proxy-00000000'
>> spce*CLI> exit
>> "
>>
>> Is this the issue that is going to be fixed in the new version, or is this just an issue with my installs? Perhaps its a codec issue, but also other keys work.
>>
>> Any further insight to this problem would be greatly appreciated.
>>
>> Thank You,
>>
>> Joe Gaylord
>> COLI Inc.
>>
>>
>> On 1/3/2019 12:18 PM, Joe Gaylord wrote:
>>> The Asterisk logs is not showing any error about not finding a file and doesn't even acknowledge the pressing of the '*' key on the phone. Voicemail access internally works by dialing 2000, but not by using the * to access from the outside.
>>>
>>> Thank You,
>>>
>>> Joe Gaylord
>>>
>>> On 12/29/2018 12:15 AM, Walter Klomp wrote:
>>>> Reading the asterisk log will give you some hints especially the digits issue which is a long standing problem. You just have to put the digit files in the right place. (Somewhere in /usr/share if my memory serves me right).
>>>>
>>>> Yours sincerely,
>>>> Walter
>>>>
>>>> On 29 Dec 2018, at 3:40 AM, Joe Gaylord <joe at torchlake.com> wrote:
>>>>
>>>>> I have the asterisks sounds, but the problem is accessing the voicemail from the outside and pressing of the asterisk key '*' doesn't interrupt the unavailable greeting. It just doesn't do anything, when pressed.
>>>>>
>>>>> Joe Gaylord
>>>>> COLI Inc.
>>>>>
>>>>> On 12/28/2018 12:17 PM, Joe Gaylord wrote:
>>>>>> Has there been any solution to this? I am having the same problem and can't seem to find a work-around.
>>>>>>
>>>>>> Thank You,
>>>>>>
>>>>>> Joe Gaylord
>>>>>> COLI Inc.
>>>>>>
>>>>>> On 3/13/2018 1:51 PM, Clint Wiley wrote:
>>>>>>> I’m having the same problems.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> _________________________
>>>>>>> Clint Wiley
>>>>>>> Hagerstown Fiber Internet
>>>>>>>
>>>>>>>> On Mar 13, 2018, at 1:42 PM, Brian Pelletier <brian.pelletier at aloe-me.net> wrote:
>>>>>>>>
>>>>>>>> Just doing a follow up to see if anyone has experienced either one of these issues or have a good starting spot for me.
>>>>>>>>
>>>>>>>> Brian Pelletier
>>>>>>>>
>>>>>>>>
>>>>>>>> ---- On Fri, 09 Mar 2018 11:36:37 -0500 <brian.pelletier at aloe-me.net> wrote ----
>>>>>>>> I am having 2 issues with voicemail since my initial install. I am running mr5.5.3.
>>>>>>>>
>>>>>>>> ISSUE 1 -
>>>>>>>> when dialing into the voicemail system or hitting someone voicemail box hitting the "*" should interrupt the message from what I have read and allow you to enter you phone number to gain access to your box. hitting the star only makes a sound and does nothing hitting "#" still skips the greeting and all other buttons work (including "*") once you are in the voicemail system. I cant find any errors in any logs and I have tried using different codex with no luck. has anyone run into this and know if its just a bug or how to go about fixing it?
>>>>>>>>
>>>>>>>> ISSUE 2 -
>>>>>>>> when trying checking voicemails from a phone by dialing in I am getting a fast busy when it gets to the point of announcing the time stamp. I hear the number of messages in the box and it starts to say to announce "message 1 loft on...), this is where it goes fast busy. The messages to email out just fine and the WAV file plays. I can even log into the CSC portal and play the messages there and delete them after. In the log I can see that the system does read correctly how many messages the box has but when it goes to play them it says no file found. I'm guessing that maybe the config file is looking in the wrong place or maybe storing them in the wrong place. Anyone know what directions i should go with this one ?
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Brian Pelletier
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
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>>>>>>>> Spce-user at lists.sipwise.com
>>>>>>>> https://lists.sipwise.com/listinfo/spce-user
>>>>>>>
>>>>>>>
>>>>>>>
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>>>>>>
>>>>>>
>>>>>> --
>>>>>> Michael 'Joe' Gaylord
>>>>>> COLI Inc.
>>>>>> Office: 888.299.0071 Ext 220
>>>>>> Office Direct: 231.354.7063
>>>>>> Fax: 888.990.0709
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Spce-user mailing list
>>>>>>
>>>>>> Spce-user at lists.sipwise.com
>>>>>> https://lists.sipwise.com/listinfo/spce-user
>>>>>
>>>>>
>>>>> --
>>>>> Michael 'Joe' Gaylord
>>>>> COLI Inc.
>>>>> Office: 888.299.0071 Ext 220
>>>>> Office Direct: 231.354.7063
>>>>> Fax: 888.990.0709
>>>>>
>>>>> _______________________________________________
>>>>> Spce-user mailing list
>>>>> Spce-user at lists.sipwise.com
>>>>> https://lists.sipwise.com/listinfo/spce-user
>>>>
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>>>
>>>
>>> --
>>> Michael 'Joe' Gaylord
>>> COLI Inc.
>>> Office: 888.299.0071 Ext 220
>>> Office Direct: 231.354.7063
>>> Fax: 888.990.0709
>>>
>>>
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>>
>>> Spce-user at lists.sipwise.com
>>> https://lists.sipwise.com/listinfo/spce-user
>>
>>
>> --
>> Michael 'Joe' Gaylord
>> COLI Inc.
>> Office: 888.299.0071 Ext 220
>> Office Direct: 231.354.7063
>> Fax: 888.990.0709
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> https://lists.sipwise.com/listinfo/spce-user
>
>
> --
> Michael 'Joe' Gaylord
> COLI Inc.
> Office: 888.299.0071 Ext 220
> Office Direct: 231.354.7063
> Fax: 888.990.0709
>
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> https://lists.sipwise.com/listinfo/spce-user
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