"press 1 for sales" doesnt work on some phones

Aug 19, 2008 at 10:17 AM
Hi Guys, i am a new user to Call butler and love its simplicity so far but have a small issue i cannot resolve on my own.

I have 2 voip lines setup and working perfectly. Ie can dial out with x-lite and and can recieve calls from mobiles and when i press 1 it transfers to x-lite and voicemail as planned.

When i dial from my office phone line or have a friend call me from his office phone line from another location the welcome and options messages work but you cannot select an option. Pressing option 1 does nothing. (should transfer to ext 101)

I have tested with IVM from NCH software and with this when i press 1 it diverts to line 1 ok so I am sure it must be a setting in call butler i have got wrong and not the hardware.

Please help if possible as I dont want to use IVM - its far too complicated compared to call bulter.

Thanks  Chris

Aug 19, 2008 at 6:42 PM
Not sure why this helps or if it even works, but everyone was saying on the old forums, when you have a dtmf problem uncheck stun in network settings.
Aug 19, 2008 at 7:41 PM
Thanks for the info - I have tried this but unfortunately still not working - ANY other ideas would be welocme as I would love to have call butler go live but it needs to support all phone lines for this.
Aug 19, 2008 at 10:32 PM
So you have 2 VOIP providers (which one)? or 2 different phones/softphones?

And the dtmf recognition works on your cellphone but not on landlines?
Aug 20, 2008 at 7:43 AM
I currently have 2 voip lines both from BT ( in the uk) . I have call butler installed and the x-lite softphone.

Before installing call butler i tried IVM from NCH and this installed and worked on both lines both calling in and out. "Press 1 for sales" etc from my office landline also worked well but it was verycomplicated to add music on hold etc , so I decided to go to a more user friendly platform - hence Call Butler. The setup seems to be right with call butler as it all works well when i dial from a mobile and some landlines ( ie pressing 1 transfers to sales ) but other landlines like the one in my office which is still a normal pstn line doesnt work ( ie You hear welcome message but pressing 1, 2 3 etc does nothing - like the key was never pressed. then after a few seconds says do you wish to continue this call )


Aug 20, 2008 at 2:20 PM
what codecs do you have checked in cb and what codecs does the voip provider support.
Aug 20, 2008 at 3:37 PM
Im Just trying to find out from BT now. I didnt realise that the voice codecs would matter and thought that this was only relevant for the local machine running CB.

Ill advise as soon as BT come back top me with the info. THANKS AGAIN
Aug 20, 2008 at 3:40 PM

Have just got the info from BT - They use G.711 and G.712 - as these are not in the CB codec list do you have any suggestions as it seems i cant add another one into CB

 

Coordinator
Aug 20, 2008 at 3:55 PM
CallButler does support G.711.  G.711 is the same thing as PCM (pulse code modulation).  There are two "flavors" of PCM - PMCu (sometimes called mu-law) and PCMa (sometimes called A-law).  PCMU is largely used in North American and Japan.  PCMa is largely used in Europe and the rest of the world.
Aug 20, 2008 at 5:01 PM
Correct me if I'm wrong anyone, but I do believe some codecs can only send dtmf inband while others can only send it rfc2833 (and then to top it off you can send it via SIP), so that is why I asked.
Aug 20, 2008 at 5:03 PM
Also, in callbutler can you uncheck everything but pcmu? then give it a try?
Aug 20, 2008 at 6:03 PM
Hi Guys , thanks for the input but sorry still no joy. I have tried every combination with both PCMu at top, PCMa at top and everything else checked - unchecked and in different orders but still no transfer when i press 1 from some phone lines.
Aug 20, 2008 at 6:32 PM
Download wireshark, capture all SIP packets, make the call, hit 1 a few times, and post the packets please.
Aug 20, 2008 at 7:33 PM
Hi, i have just downloaded and installed wire shark but cant seem to get it to work for sip packets - think im being a bit thick - be gratefeul if you could explain what settings i need. Thanks Chris
Aug 21, 2008 at 7:42 AM
I think i have this running as requested but can you advise if i need to post something different.

0000  00 50 8d db 59 84 00 1b  5b 05 ff 59 08 00 45 00   .P..Y... [..Y..E.
0010  02 45 00 00 40 00 75 11  5d 84 3e ef 0f 84 56 a5   .E..@.u. ].>...V.
0020  01 0c 13 c4 13 c4 02 31  3a 22 42 59 45 20 73 69   .......1 :"BYE si
0030  70 3a 30 35 36 30 31 32  37 39 37 35 30 40 38 36   p:056012 72750@86
0040  2e 31 36 35 2e 31 2e 31  32 3a 35 30 36 30 20 53   .165.1.1 2:5060 S
0050  49 50 2f 32 2e 30 0d 0a  56 69 61 3a 20 53 49 50   IP/2.0.. Via: SIP
0060  2f 32 2e 30 2f 55 44 50  20 36 32 2e 32 33 39 2e   /2.0/UDP  62.239.
0070  31 35 2e 31 33 32 3a 35  30 36 30 3b 62 72 61 6e   15.132:5 060;bran
0080  63 68 3d 7a 39 68 47 34  62 4b 36 67 64 75 6c 6c   ch=z9hG4 bK6gdull
0090  33 30 62 6f 67 68 72 63  63 37 36 37 34 30 63 64   30boghrc c76740cd
00a0  6c 70 67 75 75 32 30 2e  31 0d 0a 41 6c 6c 6f 77   lpguu20. 1..Allow
00b0  2d 45 76 65 6e 74 73 3a  20 6d 65 73 73 61 67 65   -Events:  message
00c0  2d 73 75 6d 6d 61 72 79  0d 0a 41 6c 6c 6f 77 2d   -summary ..Allow-
00d0  45 76 65 6e 74 73 3a 20  72 65 66 65 72 0d 0a 41   Events:  refer..A
00e0  6c 6c 6f 77 2d 45 76 65  6e 74 73 3a 20 64 69 61   llow-Eve nts: dia
00f0  6c 6f 67 0d 0a 41 6c 6c  6f 77 2d 45 76 65 6e 74   log..All ow-Event
0100  73 3a 20 6c 69 6e 65 2d  73 65 69 7a 65 0d 0a 4d   s: line- seize..M
0110  61 78 2d 46 6f 72 77 61  72 64 73 3a 20 36 39 0d   ax-Forwa rds: 69.
0120  0a 43 61 6c 6c 2d 49 44  3a 20 53 44 6f 32 38 37   .Call-ID : SDo287
0130  38 30 31 2d 33 36 39 30  35 35 66 66 35 33 33 30   801-3690 55ff5330
0140  34 62 31 63 65 36 32 30  36 62 66 39 37 39 65 30   4b1ce620 6bf979e0
0150  30 61 34 64 2d 75 70 72  76 30 32 32 0d 0a 46 72   0a4d-upr v022..Fr
0160  6f 6d 3a 20 3c 73 69 70  3a 30 31 39 35 32 36 31   om: <sip :0197261
0170  37 38 30 30 40 62 6d 6e  68 62 2d 30 32 2e 62 74   7800@bmn hb-02.bt
0180  2e 63 6f 6d 3b 74 72 61  6e 73 70 6f 72 74 3d 75   .com;tra nsport=u
0190  64 70 3e 3b 74 61 67 3d  53 44 6f 32 38 37 38 30   dp>;tag= SDo28780
01a0  31 2d 62 74 2e 63 6f 6d  2b 31 2b 33 31 62 64 31   1-bt.com +1+31bd1
01b0  2b 65 34 34 33 37 63 34  0d 0a 54 6f 3a 20 3c 73   +e4437c4 ..To: <s
01c0  69 70 3a 30 35 36 30 31  32 37 39 37 35 30 40 62   ip:05601 272750@b
01d0  6d 6e 68 62 2d 30 32 2e  62 74 2e 63 6f 6d 3e 3b   mnhb-02. bt.com>;
01e0  74 61 67 3d 36 43 39 42  34 42 43 34 44 33 31 41   tag=6C9B 4BC4D31A
01f0  34 43 35 45 38 32 35 31  32 35 37 31 38 34 35 35   4C5E8251 25718455
0200  36 32 37 36 0d 0a 43 53  65 71 3a 20 38 39 33 36   6276..CS eq: 8936
0210  34 34 35 34 39 20 42 59  45 0d 0a 4f 72 67 61 6e   44549 BY E..Organ
0220  69 7a 61 74 69 6f 6e 3a  20 0d 0a 53 75 70 70 6f   ization:  ..Suppo
0230  72 74 65 64 3a 20 31 30  30 72 65 6c 0d 0a 43 6f   rted: 10 0rel..Co
0240  6e 74 65 6e 74 2d 4c 65  6e 67 74 68 3a 20 30 0d   ntent-Le ngth: 0.
0250  0a 0d 0a                                           ...    
Aug 22, 2008 at 1:30 AM
I'm not interested in the HEX equivalent of the plain text SIP.  Im looking for something like this:  You get the following from stats -> voip calls -> graph.  And then what would be even better yet hit player and listen for the dtmf tones you hit to see if you can hear them.  This way we will be killing 2 dtmf transports in 1 via sip and inband.  Then do it from a phone that works and see if you see a change.

|Time     | 10.10.10.6        | 10.10.9.1         |
|148.009  |         INVITE SDP ( g711U g711A g729 telephone-event)          |SIP From: sip:108@10.10.9.1 To:sip:##########@10.10.9.1
|         |(5060)   ------------------>  (5060)   |
|148.010  |         407 Proxy Authentication Required          |SIP Status
|         |(5060)   <------------------  (5060)   |
|148.026  |         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
|148.030  |         INVITE SDP ( g711U g711A g729 telephone-event)          |SIP From: sip:108@10.10.9.1 To:sip:##########@10.10.9.1
|         |(5060)   ------------------>  (5060)   |
|148.031  |         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|152.270  |         180 Ringing                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|152.271  |         183 Session Progress SDP ( g711U g711A telepho...event)          |SIP Status
|         |(5060)   <------------------  (5060)   |
|152.462  |         RTP (g711U)                   |RTP Num packets:711  Duration:14.200s SSRC:0x1D16DDF
|         |(2224)   ------------------>  (14292)  |
|152.948  |         RTP (g711U)                   |RTP Num packets:1591  Duration:31.799s SSRC:0x24A0555E
|         |(2224)   <------------------  (14292)  |
|166.676  |         200 OK SDP ( g711U g711A telephone-event)          |SIP Status
|         |(5060)   <------------------  (5060)   |
|166.683  |         RTP (g711U)                   |RTP Num packets:905  Duration:18.080s SSRC:0x1D16DDF
|         |(2224)   ------------------>  (14292)  |
|166.693  |         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
Aug 22, 2008 at 1:42 AM

but obviously we need the one from cb to voip provider not phone to cb...
Aug 22, 2008 at 1:46 AM
Alpha maybe able to clarify or tell me im going in the wrong direction, but what I think is happening is your VOIP provider has more then one gateway to PSTN, some probably use one kind of dtmf and the other use anohter kind, that's why when you call from some phones it works and others it doesn't.  And I do not think cb supports all 3 dtmf transports, again alpha correct me if im wrong...
Aug 22, 2008 at 8:13 PM
http://www.codeplex.com/callbutler/Thread/View.aspx?ThreadId=33738 well this is a much easier way that mike explains...I didn't know call butler had this built in :)
Coordinator
Aug 22, 2008 at 8:41 PM
Yes we do support all three DTMF transports, RTP (Out of Band), SIP INFO and In-Band
Aug 22, 2008 at 9:07 PM
Any other ideas why the dtmf would work on some incomming calls and not others then?
Coordinator
Aug 27, 2008 at 6:00 PM
I guess the first question would be:

Is there a firewall issue with the phone that isn't working?

It's highly possible that other than SIP messages, CallButler can't "hear" incoming audio packets from the other phone. Is the phone that's not working an IP phone? If possible on the phone that isn't working, can you try changing the DTMF transport to SIP info?
Aug 27, 2008 at 6:29 PM
Hi Guys - Thanks for all the info so far - i will try the doagnostics as son as i get a minute but as im running a small business i dont even get enough time to sleep at the moment.

I have turned off ALL firewalls virus softeware etc and the issue still persists.  These are calls coming in from a standard phone line with PSTN connection. They are not ip phones.

Therefore I assume i cant change DTMF transport to SIP info.

I have re installed NCH IVM and this work fine with the phones that dont work. Is there anything else i can do as i would far rather use CB . Its much much sleeker if only it could recognise the call tones.

Thanks again

Coordinator
Aug 28, 2008 at 6:02 PM
Which VoIP provider are you using? In many cases you can have them change the way DTMF tones are sent to us. Also, are you outside of the U.S. by any chance?
Aug 29, 2008 at 9:36 AM
I amn using BT and i am based in the UK. Dont suppose you have a version that will work here ..... Cheers
Coordinator
Aug 29, 2008 at 4:33 PM
Connexin is a great UK VoIP provider http://www.connexin.co.uk/ that works perfectly with CallButler
Aug 29, 2008 at 5:12 PM
Hi Mikey, thanks for the info. I am currenmtly signed up with BT and have a 12 month contract left to go. What i can understand is why NCH IVM works but call bultler doesnt. I was assuming it was something i have set wrong in call butler.
Coordinator
Aug 29, 2008 at 5:19 PM
cap,

Is there any setting with BT where you can change the way they send you DTMF tones? Some VoIP providers will allow you to do this.

Another thing I would do is maybe try to find another CallButler user in the UK. If we can find one (which I know we have a few), I would suggest you call them with the same phone and see if this just happens to be a general problem with CallButler recognizing tones from this particular phone, or if it's just specific to your system.

Mikey,

Do you think you could help cap find an existing user in the UK?

-jim
Aug 30, 2008 at 9:31 AM
Hi Alpha, I will try to find someone in the UK to see if anyone else has this working. BT tell us that they dont change the way they send DTMF tones so no luck here, however I still dont understand this because IVM from NCH works fine which suggestes the VOIP is ok and that there is something within the CB software that isnt working in the UK.
Sep 1, 2008 at 7:57 AM
Hi Guys, As you seem to think the VOIP provider maybe an issue inbouns as the outbound works fine do you think a Grandstream voip gateway would solve the problems. I have an incomming PSTN line at the moment that I was going to get forwarded to VOIP so running this through a voip gateway myself would save money in the longrun and if works would allow me to use call butler.