[asterisk-bugs] [JIRA] (ASTERISK-20975) DTMF issue with SIP trunk
Michael L. Young (JIRA)
noreply at issues.asterisk.org
Wed Jan 23 11:00:58 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-20975?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Michael L. Young updated ASTERISK-20975:
----------------------------------------
Description:
DTMF seems not work reliably for incoming calls through Skype Connect trunk.
The system is installed from FreePBX distro. Asterisk version is 1.8.8.0
There is a Skype connect trunk configured for LD calls. The problem is when users call a Skype number that leads to IVR, and try to reach any extension, Asterisk ignores DTMF tones.
relaxdtmf=yes
DTMF mode is RFC2833 (as Skype recommends)
jitterbuffer is set to yes
and that's what I see in logs.
{noformat}
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '8' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '8' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end '8' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end passthrough '8' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '9' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '9' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '9' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '9' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin '4' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin ignored '4' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '4' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '4' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin '3' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin ignored '3' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end '3' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end passthrough '3' on SIP/SkypeConnect-00000652
{noformat}
The duration seems to be ok.
But what's about this DTMF ignoring?
was:
DTMF seems not work reliably for incoming calls through Skype Connect trunk.
The system is installed from FreePBX distro. Asterisk version is 1.8.8.0
There is a Skype connect trunk configured for LD calls. The problem is when users call a Skype number that leads to IVR, and try to reach any extension, Asterisk ignores DTMF tones.
relaxdtmf=yes
DTMF mode is RFC2833 (as Skype recommends)
jitterbuffer is set to yes
and that's what I see in logs.
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '8' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '8' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end '8' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end passthrough '8' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '9' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '9' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '9' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '9' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin '4' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin ignored '4' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '4' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '4' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin '3' received on SIP/SkypeConnect-00000652
[2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin ignored '3' on SIP/SkypeConnect-00000652
[2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end '3' received on SIP/SkypeConnect-00000652, duration 240 ms
[2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end passthrough '3' on SIP/SkypeConnect-00000652
The duration seems to be ok.
But what's about this DTMF ignoring?
> DTMF issue with SIP trunk
> -------------------------
>
> Key: ASTERISK-20975
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20975
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Environment: Cent OS 5.8
> Reporter: Anna Vlasenko
>
> DTMF seems not work reliably for incoming calls through Skype Connect trunk.
> The system is installed from FreePBX distro. Asterisk version is 1.8.8.0
> There is a Skype connect trunk configured for LD calls. The problem is when users call a Skype number that leads to IVR, and try to reach any extension, Asterisk ignores DTMF tones.
> relaxdtmf=yes
> DTMF mode is RFC2833 (as Skype recommends)
> jitterbuffer is set to yes
> and that's what I see in logs.
> {noformat}
> [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '8' received on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '8' on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end '8' received on SIP/SkypeConnect-00000652, duration 240 ms
> [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF end passthrough '8' on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin '9' received on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:39] DTMF[13332] channel.c: DTMF begin ignored '9' on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '9' received on SIP/SkypeConnect-00000652, duration 240 ms
> [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '9' on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin '4' received on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF begin ignored '4' on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end '4' received on SIP/SkypeConnect-00000652, duration 240 ms
> [2013-01-23 16:09:40] DTMF[13332] channel.c: DTMF end passthrough '4' on SIP/SkypeConnect-00000652 [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin '3' received on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF begin ignored '3' on SIP/SkypeConnect-00000652
> [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end '3' received on SIP/SkypeConnect-00000652, duration 240 ms
> [2013-01-23 16:09:41] DTMF[13332] channel.c: DTMF end passthrough '3' on SIP/SkypeConnect-00000652
> {noformat}
> The duration seems to be ok.
> But what's about this DTMF ignoring?
>
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