[Spce-user] Voicemail - '*' interrupt not working and message not playing from phone
Joe Gaylord
joe at torchlake.com
Thu Jan 10 09:52:02 EST 2019
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
> <mailto: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
>>> <mailto: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
>>>>>>> <mailto: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
>>>>>>> <mailto: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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Spce-user mailing list
>>>>>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
>>>>>>> https://lists.sipwise.com/listinfo/spce-user
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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 <mailto:Spce-user at lists.sipwise.com>
>>>> https://lists.sipwise.com/listinfo/spce-user
>>>
>>> The contents of this email and any attachments are confidential and
>>> may also be privileged. You must not disseminate the contents of
>>> this email and any attachments without permission of the sender. If
>>> you have received this email by mistake, please delete all copies
>>> and inform the sender immediately. You may refer to our company's
>>> Privacy Policy here
>>> <https://myrepublic.net/sg/legal/terms-of-use-policies/privacy-policy/>.
>>
>>
>>
>> --
>> 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 <mailto: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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20190110/cffa4f28/attachment-0001.html>
More information about the Spce-user
mailing list