Inbound no longer accepts # or * to retrieve voicemail

Unanswered Question
Oct 2nd, 2009

I have a strange problem that cropped up in my customer’s uc520. I had 8 PSTN lines and a working sip trunk. I had the dial plan set to add a 1 to the number being dialed. All the user had to enter was 9 for an outside line and the area code and number. I ported all 8 analog lines to the sip trunk provider. After the numbers ported everything was working fine.  I even installed and started using CCA 2.1 .After about two weeks I decided to unplug the PSTN lines from the uc520, as they were no longer hooked up. After I did that people could no longer dial without the 1. They now had to enter 9 + 1 + area code + number. I tried to change the outbound dial plan but no matter what I did it would not work. I put the dial plan back to using the 1 and now it works, but they still have to dial a 1. Then on Wednesday of this week I found out that dialing in to retrieve voicemail was not working. When the voicemail asked for user id followed by the # sign it would not recognize the # or * key. This system has had a lot of customizing by both TAC and me, so I don’t know if CCA 2.1 broke something or if there is another issue. The uc520 sits behind a sr520 which is connected to the internet via a Comcast cable line. I have launched a ticket with the sip provider, but they will probably come back and say that it has nothing to do with them, (as usual).

I have this problem too.
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Maulik Shah Fri, 10/02/2009 - 15:07

1. Hard to follow the logs - but I see a few things that are not quite right:

- INVITE for outbound call from UC500 uses RTP payload type 101 for DTMF (RFC2833)

m=audio 19396 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

- 200 OK from SP has RTPpayload type 100 and events 0-15, not 0-16

m=audio 52044 RTP/AVP 18 0 100


a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15

You want to change the payload type on the UC500 to use 100 - see section 4.4.11 on link below:

2. Do * / # work internally from an IP phone to voicemail?

3. Can you gather a clean log with deb ccsip mess for one failed call

Barry Hunsinger Wed, 10/07/2009 - 07:34

when I dial the voicemail extension directly (200) the # works, but it then asks you to press * if you have a mailbox on the system, but * is not recognized. also I have the zero out in voicemail set to goto the autoattendant (300) when you are in someones voicemail to get out of voicemail and into the autoattendant. This has stopped working from the outside. I will try to get better logs.

Barry Hunsinger Wed, 10/07/2009 - 08:06

I enabled the troubleshooting logs for sip. I then made a call from my cell phone (908-265-2006) to our sip line (908-753-0800). when the AA answered I pressed 225 (my Extension) to get to my voicemail . while in voicemail I pressed 0 to zero out to the same AA. nothing happened. I then recalled (same numbers) and pressed #, the AA said to try again. I then pressed the * key and nothing happened. Icalled a third time and in the AA I pressed 7 which brings up the prompt to enter your user id. It then askes for your id followed by #. it appears to accept the number, but not the #. it says it could not hear the entry.  I have attached the logs. Should I open a TAC case?

Maulik Shah Wed, 10/07/2009 - 12:22

A TAC case would be best so this can be looked at in real time. Couple of comments that you may want to clarify for the TAC engineer:

1. Does DTMF work reliably for all digits from an internal extension to the AA - i.e. you can press 0-9,#,* and no issues seen - correct?

2. If the DTMF is unreliable only for calls over the SIP Trunk, then a few things to check:

a. What DTMF method does your provider use - it looks to be RFC2833 and as I mentioned earlier you may need to change the payload type to match

b. The logs on CCA are missing one debug option (we will add this in the next release) - to look at this issue you need th ebelow debugs for one call only:

deb ccsip all

deb voip rtp session named-event

deb voip ccapi inout

Would need to check what is received from the SIP Trunk provider first and then what gets sent to the AA.


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