Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. Unfortunately I can't do that - but looking at the captures I can see some slight differences between working and non working scenarios: http://fotobytes.co.uk/user22171/dtmf_debug.php Perhaps you can suggest where I should next look to trouble shoot this? You can try the rfc2833compensate option... Other than that I can't know until I see a packet capture. Tried, but this was not successful :-( What I've done is detailed here: http://fotobytes.co.uk/user22171/dtmf_debug.php Once I get to the bottom of it, I'll write it up properly - for the benefit of other Sipgate/Asterisk users. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
listu...@spamomania.co.uk wrote:
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the general section of sip.conf
This has made no difference. I've tried a range of settings (auto,
rfc2833,info) but no matter what, it plain refuses to pick up key
presses.
Locally, if I call from an extension on an ata or a softphone, it works
flawlessly (I have no fxo, everything is SIP based).
It's extremely frustrating and I would be grateful if anyone could offer
some help troubleshooting and fixing this?
Hi,
maybe your RTP stream is not getting through the asterisk box due to
canreinvite=yes setting in your SIP profile?
What is result of the following test in your dialplan?
exten = 123,1,NoOp(***INCOMING CALL***)
exten = 123,n,Set(CHANNEL(language)=en)
exten = 123,n,Answer()
exten = 123,n,Read(CONFNO,conf-getconfno,4)
exten = 123,n,Playback(conf-enteringno)
exten = 123,n,SayDigits(${CONFNO})
exten = 123,n,Hangup
Cheers
Joern
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Re: [asterisk-users] Sipgate DTMF not detected
On Tue, 2010-01-19 at 13:15 +0100, joern wrote:
listu...@spamomania.co.uk wrote:
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the general section of sip.conf
This has made no difference. I've tried a range of settings (auto,
rfc2833,info) but no matter what, it plain refuses to pick up key
presses.
Locally, if I call from an extension on an ata or a softphone, it works
flawlessly (I have no fxo, everything is SIP based).
It's extremely frustrating and I would be grateful if anyone could offer
some help troubleshooting and fixing this?
Hi,
maybe your RTP stream is not getting through the asterisk box due to
canreinvite=yes setting in your SIP profile?
Nope :-(
less /etc/asterisk/sip.conf | grep canreinvite
canreinvite=no
canreinvite=no
canreinvite=no
canreinvite=no
canreinvite=no
canreinvite=no
What is result of the following test in your dialplan?
exten = 123,1,NoOp(***INCOMING CALL***)
exten = 123,n,Set(CHANNEL(language)=en)
exten = 123,n,Answer()
exten = 123,n,Read(CONFNO,conf-getconfno,4)
exten = 123,n,Playback(conf-enteringno)
exten = 123,n,SayDigits(${CONFNO})
exten = 123,n,Hangup
As per my current problem.
SIPGATE CUSTOMER - SIPGATE - ASTERISK {WORKS}
@inbound-sipgate-584e;2 Playing 'conf-enteringno.gsm' (language
'en')
/PRESS 1234 AND READ BACK DETECTED WITHOUT ERROR/
-- Executing [...@cc-test:6] SayDigits(@inbound-sipgate-584e;2,
1234) in new stack
BUT - PSTN - SIPGATE is nogo:
PSTN - SIPGATE - ASTERISK {BROKEN}
SIP/sipgate-dv-0008 Playing 'conf-getconfno.gsm' (language 'en')
/MASH KEYS AS MANY TIMES AS YOU LIKE - NOTHING DETECTED/
-- User disconnected
Cheers
Joern
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Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Thanks for that. Looking at the RTP packets I can see two types as you point out. The first appears to be the audio: Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) Payload type: ITU-T G.711 PCMU (0) And as you say, the DTMF events are clear to see: RFC 2833 RTP Event Event ID: DTMF One 1 (1) ..00 1010 = Volume: 10 So, as these can be seen in the stream, do I need to tell Asterisk to detect these? Does it not do that when I set: dtmfmode=rfc2833 ??? There are some pretty widely recognized RFC2833 compatibility issues in the SIP/RTP world. Which version of Asterisk are you using? Do you know what kind of equipment your carrier is using? If they are using Asterisk you can try to add rfc2833compensate=yes to their peer entry in sip.conf. The SIP debug, however, will tell you if the remote end is configured to use RFC2833 or not. That's why I was telling you to look for telephone-event in the INVITE from your provider. Keep in mind SIP (most likely) runs over UDP between you and your provider, not TCP. I see a 'telephone-event' : a=rtpmap:101 telephone-event/8000 That's all you need to know. They are configured for RFC2833 and they're sending RFC2833. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 10:12 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: Thanks for that. Looking at the RTP packets I can see two types as you point out. The first appears to be the audio: Real-Time Transport Protocol 10.. = Version: RFC 1889 Version (2) Payload type: ITU-T G.711 PCMU (0) And as you say, the DTMF events are clear to see: RFC 2833 RTP Event Event ID: DTMF One 1 (1) ..00 1010 = Volume: 10 So, as these can be seen in the stream, do I need to tell Asterisk to detect these? Does it not do that when I set: dtmfmode=rfc2833 ??? There are some pretty widely recognized RFC2833 compatibility issues in the SIP/RTP world. I had a nasty feeling something like that was coming :-( Which version of Asterisk are you using? Asterisk 1.6.1.11 Do you know what kind of equipment your carrier is using? If they are using Asterisk you can try to add rfc2833compensate=yes to their peer entry in sip.conf. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] The SIP debug, however, will tell you if the remote end is configured to use RFC2833 or not. That's why I was telling you to look for telephone-event in the INVITE from your provider. Keep in mind SIP (most likely) runs over UDP between you and your provider, not TCP. I see a 'telephone-event' : a=rtpmap:101 telephone-event/8000 That's all you need to know. They are configured for RFC2833 and they're sending RFC2833. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? You can try the rfc2833compensate option... Other than that I can't know until I see a packet capture. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote: On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: ..snip.. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources. [http://www.sipgate.co.uk/faq/index.php?do=displayArticlearticle=540qw=asterisk] Just because they don't offer assistance with Asterisk doesn't mean they don't use it themselves. If you send me a packet capture in PCAP format with SIP+RTP between your system and your carrier I can debug this further. That may contain sensitive data, such as SIP account/password details - so I'll pass on that, but thanks for the offer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Wed, Jan 13, 2010 at 12:07 PM, listu...@spamomania.co.uk listu...@spamomania.co.uk wrote: That may contain sensitive data, such as SIP account/password details - so I'll pass on that, but thanks for the offer. Even if they are using auth it's challenge response and fairly difficult to reverse engineer, not that I have the time for that. I do however, specialize in debugging DTMF. I always make time for interesting cases. I also own a voice service provider so it's unlikely I'm interested in your sipgate credentials :). Good luck. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net wrote: Codec? I've had 2833 do funny things with anything other than a/ulaw (might just be me though..) S -- Codecs other than G711u/a don't support inband DTMF. Seeing as INFO is rarely used that pretty much leaves RFC2833. Turn on SIP debugging and look in the INVITE from the provider for telephone-event. If you see it, they're configured to use RFC2833. If they are, you need to do a packet capture or other RTP debug to see the out of band RFC2833 events. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote:
On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes steve-li...@geekinter.net
wrote:
Codec? I've had 2833 do funny things with anything other than a/ulaw
(might just be me though..)
S
--
Codecs other than G711u/a don't support inband DTMF. Seeing as INFO
is rarely used that pretty much leaves RFC2833. Turn on SIP debugging
and look in the INVITE from the provider for telephone-event. If you
see it, they're configured to use RFC2833.
If they are, you need to do a packet capture or other RTP debug to see
the out of band RFC2833 events.
--
Kristian Kielhofner
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
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Re: [asterisk-users] Sipgate DTMF not detected
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
RFC2833 is carried in RTP like the audio stream. However, it uses a
different payload type from the RTP packets used to transport the
audio. If you did an RTP capture you would be able to see the RFC2833
events (which correspond to DTMF presses).
The SIP debug, however, will tell you if the remote end is configured
to use RFC2833 or not. That's why I was telling you to look for
telephone-event in the INVITE from your provider. Keep in mind SIP
(most likely) runs over UDP between you and your provider, not TCP.
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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Re: [asterisk-users] Sipgate DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]
On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote:
On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Assuming that I enable debugging using:
asterisk -rvv
CLI sip set debug on
Then with this:
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
dump shows nothing. SIPGATE have said;
you should be able to set the dtmfmode to rfc2833 in your default
sip.conf.
Best regards,
Frederik
I've tried other combinations such as info, inband et al. I'm guessing
{that's all it is} that rfc2833 will signal the dtfm over sip as opposed
to in the audio stream?
RFC2833 is carried in RTP like the audio stream. However, it uses a
different payload type from the RTP packets used to transport the
audio. If you did an RTP capture you would be able to see the RFC2833
events (which correspond to DTMF presses).
Thanks for that. Looking at the RTP packets I can see two types as you
point out. The first appears to be the audio:
Real-Time Transport Protocol
10.. = Version: RFC 1889 Version (2)
Payload type: ITU-T G.711 PCMU (0)
And as you say, the DTMF events are clear to see:
RFC 2833 RTP Event
Event ID: DTMF One 1 (1)
..00 1010 = Volume: 10
So, as these can be seen in the stream, do I need to tell Asterisk to
detect these? Does it not do that when I set: dtmfmode=rfc2833
???
The SIP debug, however, will tell you if the remote end is configured
to use RFC2833 or not. That's why I was telling you to look for
telephone-event in the INVITE from your provider. Keep in mind SIP
(most likely) runs over UDP between you and your provider, not TCP.
I see a 'telephone-event' :
a=rtpmap:101 telephone-event/8000
buried in the chunk below. but I have to be honest, SIP is new to me so
I'm not sure of myself with this:
v=0
o=root 27089 27089 IN IP4 217.10.69.13
s=session
c=IN IP4 217.10.69.13
t=0 0
m=audio 19990 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
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Re: [asterisk-users] Sipgate DTMF not detected
On 11 Jan 2010, at 16:26, listu...@spamomania.co.uk wrote: This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. snip It's extremely frustrating and I would be grateful if anyone could offer some help troubleshooting and fixing this? Try asking Sipgate what settings you should use? If they are sending it as audio, make sure you are using suitable codecs etc. Try SIP traces to see what you can see. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On Mon, 2010-01-11 at 16:52 +, Steve Howes wrote: On 11 Jan 2010, at 16:26, listu...@spamomania.co.uk wrote: This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. snip It's extremely frustrating and I would be grateful if anyone could offer some help troubleshooting and fixing this? Try asking Sipgate what settings you should use? If they are sending it as audio, make sure you are using suitable codecs etc. Try SIP traces to see what you can see. Steve Steve, you've snipped the bit that said: There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf But thanks, been there and done that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipgate DTMF not detected
On 11 Jan 2010, at 17:06, listu...@spamomania.co.uk wrote: On Mon, 2010-01-11 at 16:52 +, Steve Howes wrote: Try asking Sipgate what settings you should use? If they are sending it as audio, make sure you are using suitable codecs etc. Try SIP traces to see what you can see. Steve, you've snipped the bit that said: There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf But thanks, been there and done that. Codec? I've had 2833 do funny things with anything other than a/ulaw (might just be me though..) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
