[asterisk-users] atx timeout - play xferfailsound
Asterisk 1.6.2.20 on Debian Lenny I'm finding that if no one answers an attended transfer (timeout set by atxfernoanswertimeout), then the transferrer is handed back to the original caller, and a beep is played. In 1.4 I was able to indicate the timeout and failure by setting xferfailsound to a custom recording, but this doesn't seem to happen in 1.6 How can I indicate a timeout to the transferrer? Many thanks John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and configuration to be via Database instead of conf files
Dear Binni; My asterisk version is: Connected to Asterisk 1.4.39.1-vici RPM by dem...@goautodial.com So it is only by 1.4.19? By the way, the version I am using has been installed using goautodial. Regards Bilal Hi, I've played around with using a database configuration for Asterisk and it certainly works in 1.4.19. If you want any help setting up the configuration you can contact me directly. Regards Binni -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User hit f to disconnect call.
--- On Thu, 1/26/12, Kevin P. Fleming kpflem...@digium.com wrote: From: Kevin P. Fleming kpflem...@digium.com Subject: Re: [asterisk-users] User hit f to disconnect call. To: asterisk-users@lists.digium.com Date: Thursday, January 26, 2012, 10:58 AM On 01/26/2012 07:22 AM, Vieri wrote: Hi, I was receiving fax calls just fine until recently. I'm now having random disconnections. Faxes are received over an ISDN BRI line and Asterisk 1.4 detects it and sends it to a iaxmodem (exten 10025 below). All's apparently as expected except for the fact that the following message comes up in the Asterisk log: User hit f to disconnect call. The iaxmodem log also shows a premature hangup (see below). I did a test fax call but I certainly didn't press any key to abort the call. What does that message mean? Asterisk log (0X is destination, Y is sending fax machine): [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [fax@from-pstn-deviate-custom:12] Dial(mISDN/6-u22326, IAX2/10025/0971847022|20|d) in new stack [Jan 26 13:46:13] DEBUG[619] chan_iax2.c: prepending 8 to prefs [Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Call accepted by 127.0.0.1 (format alaw) [Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Format for call is alaw [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Called 10025/0X [Jan 26 13:46:13] VERBOSE[619] logger.c: -- IAX2/10025-3460 is ringing [Jan 26 13:46:13] VERBOSE[619] logger.c: -- User hit f to disconnect call. [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Hungup 'IAX2/10025-3460' [Jan 26 13:46:13] VERBOSE[619] logger.c: == Spawn extension (from-pstn-deviate-custom, f, 0) exited non-zero on 'mISDN/6-u22326' [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [h@from-pstn-deviate-custom:1] Macro(mISDN/6-u22326, hangupcall) in new stack 'f' is the fake DTMF control frame used inside Asterisk to indicate that a CNG tone was detected. Do you have 'faxdetect' enabled on the mISDN channel driver for that BRI? Even if you do, though, I don't know why receiving an 'f' would disconnect the call, unless you've provided the 'd' option to app_dial. Even if you did, app_dial should be smart enough to not treat 'f' as a DTMF key, but it's not (at least not in Asterisk 1.4, this may have changed in later versions). That could be it. misdn has fax detection for incoming. app_dial IS using the 'd' option. I will try not to use it. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vigor 2920 problems
Thanks for help- suggestion fixed the issue John On 21 November 2011 11:25, John Taylor j...@vetsurgeon.org.uk wrote: Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will get permission to try new firmware later! JT On 21 November 2011 10:45, Arthur Stanfield a...@dmcip.com wrote: Hi John, We've had similiar issues with customers behind the 2920 connecting to a hosted asterisk system. If you rebooted a phone it often didn't re-register, Checking the NAT sessions table on the router revealed stale nat sessions open for the phone. On the advice of Dreytek we found a fix by lowering the NAT session timeout from the default of 24hrs down to 5 minutes and installing the latest release of the firmware (3.3.7) it may not be available on the UK Site at the moment (It wasn't when we did the upgrade!) but it can be got from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/ It may help, It may not - But its quick easy fix if it does. Regards, AJ. - Original Message - From: John Taylor j...@vetsurgeon.org.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 21 November, 2011 10:20:14 AM Subject: [asterisk-users] vigor 2920 problems One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found
On 01/28/2012 10:22 AM, Din Assegaf wrote: Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8 101 [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:9029 process_sdp: Failing due to no acceptable offer found the strange thing is when using asterisk 1.6, is normal, when using asterisk 1.8.x and using another client like Ekiga is normal too, The error message is misleading; you are having this problem because the 'm' line in the SDP with the 'audio' offer has a port number of 0 (zero)., which means it is not an active media stream offer. It does not make any sense for the SDP in an INVITE for a new call to have an m-line with a port number of zero. I'll improve the error message so that this sort of situation won't be as confusing in the future. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendFax not sending AMI events
On 01/29/2012 02:34 PM, Mike Diehl wrote: On Sunday 29 January 2012 8:27:30 am Olivier wrote: 2012/1/29 Mike Diehlmdi...@diehlnet.com Hi all, I'm working with the Digium fax for Asterisk product, which is working pretty reliably for me. However, the sendfax application isn't sending status events to AMI. The receivefax application does. For example, with this call file: Channel: sip/000E0827BD1C-1 Application: sendfax Data: /tmp/voice11-7449.tiff,zfds WaitTime: 90 All I get are the call progress events. While the fax is being received by the device, the AMI is silent. The Asterisk console displays a lot of progress information. I'm expecting to get at least FaxStatus, SendFAXStatus and FaxDocument events. Any ideas? Which asterisk version are you using ? Sorry. You'd think by now, I'd know to include that. ;^) Asterisk 1.6.2.9 Planning an upgrade to 1.8, soon, though. Please open an issue with Digium Support so we can get this tracked down. Thanks. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
ChanSpy is not completely working for me.
Dialplan :
/exten = _*XXX***,n,ChanSpy(${SIPACC}) ; var $SIPACC has SIP peer
account name/
Verbose :
/[Jan 30 16:25:47] -- Executing [*204***@from-ITEL:10]
ChanSpy(SIP/itel0-2f21, itel1) in new stack
[Jan 30 16:25:48] -- SIP/itel0-2f21 Playing 'beep.alaw'
(language 'nl')/
But the spying IP-phone itel0 does not hear a thing. It should here the
conversation between SIP/itel1-2f10and SIP/ITELin-2f0d
These are the 2 channels which are talking to each other :
/SIP/itel1-2f10 SIP/ITELin-2f0d/
Any idea which setting I'm missing ?
Kind regards,
Jonas.
On 01/25/2012 11:10 AM, Ishfaq Malik wrote:
I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio coming in and out of the channel
being spied on.'
Have you tried giving it a go?
-= Info about application 'ChanSpy' =-
[Synopsis]
Listen to a channel, and optionally whisper into it.
[Description]
This application is used to listen to the audio from an Asterisk
channel.
This includes the audio coming in and out of the channel being spied
on.
If the 'chanprefix' parameter is specified, only channels beginning with
this
string will be spied upon.
While spying, the following actions may be performed:
- Dialing '#' cycles the volume level.
- Dialing '*' will stop spying and look for another channel to spy on.
- Dialing a series of digits followed by '#' builds a channel name to
append
to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing
the
digits '1234#' while spying will begin spying on the channel
'Agent/1234'.
Note that this feature will be overridden if the 'd' option is used
NOTE: TheX option supersedes the three features above in that if a
valid
single digit extension exists in the correct context ChanSpy will exit
to
it. This also disables choosing a channel based on 'chanprefix' and a
digit
sequence.
[Syntax]
ChanSpy([chanprefix][,options])
[Arguments]
options
b: Only spy on channels involved in a bridged call.
B: Instead of whispering on a single channel barge in on both
channels
involved in the call.
c(digit):
digit - Specify a DTMF digit that can be used to spy on the
next available channel.
d: Override the typical numeric DTMF functionality and instead use
DTMF to switch between spy modes.
4 - spy mode
5 - whisper mode
6 - barge mode
e(ext): Enable *enforced* mode, so the spying channel can only
monitor
extensions whose name is in theext : delimited list.
E: Exit when the spied-on channel hangs up.
g(grp):
grp - Only spy on channels in which one or more of the groups
listed ingrp matches one or more groups from the ${SPYGROUP}
variable set on the channel to be spied upon.
NOTE: bothgrp and ${SPYGROUP} can contain either a single group
or a colon-delimited list of groups, such as
'sales:support:accountin
g'.
n([mailbox][@context]): Say the name of the person being spied on
if that person has recorded his/her name. If a context is specified,
then
that voicemail context will be searched when retrieving the name,
otherwise
the 'default' context be used when searching for the name (i.e. if
SIP/1000
is the channel being spied on and no mailbox is specified, then
'1000'
will be used when searching for the name).
o: Only listen to audio coming from this channel.
q: Don't play a beep when beginning to spy on a channel, or speak
the selected channel name.
r([basename]): Record the session to the monitor spool directory.
An optional base for the filename may be specified. The default is
'
chanspy'.
s: Skip the playback of the channel type (i.e. SIP, IAX, etc) when
speaking the selected channel name.
S: Stop when no more channels are left to spy on.
v([value]): Adjust the initial volume in the range from '-4' to
'4'. A negative value refers to a quieter setting.
w: Enable 'whisper' mode, so the spying channel can talk to the
spied-on channel.
W: Enable 'private whisper' mode, so the spying channel can talk
to the spied-on channel but cannot listen to that channel.
x(digit):
digit - Specify a DTMF digit that can be used to exit the
application.
X: Allow the user to exit ChanSpy to a valid single digit numeric
extension in the current context or the context specified by the
${SP
Y_EXIT_CONTEXT} channel variable. The name of the last channel that
was
spied on will be stored in the ${SPY_CHANNEL} variable.
On Wed, 2012-01-25 at 10:15 +0100, Jonas Kellens wrote:
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012
[asterisk-users] CA Issued Certificates / TLS + SRTP
Hi all, Firstly, apologies if the answer to this question should be obvious. I have just started working with SRTP and had a self-signed certificate working perfectly. I have now purchased a CA signed certificate but can't get it to work properly with Asterisk. I think I have a configuration error. The certificate is a GeoTrust Rapid SSL certificate. I have received the my server specific crt file and also an intermediate certificate. I am not sure of the following and would greatly appreciate if someone could give me some guidance: * Can I specify the intermediate and .crt files separately in the sip.conf file? (I am thinking of a process similar to Apache where you specify three different files; server specific certificate, chain file and key file.) * Should the intermediate and server specific certificates be combined into one certificate file? * And, is it necessary to use both my server specific certificate and the intermediate certificate on the telephones or will the telephones only require the server specific certificate? My test phone is a Yealink T28P. Many thanks. Stuart -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy : how to know channel name ?
Try
exten = _*XXX***,n,ChanSpy(SIP/${SIPACC}) ; var $SIPACC has SIP peer
account name
Ish
On Mon, 2012-01-30 at 17:04 +0100, Jonas Kellens wrote:
Hello,
ChanSpy is not completely working for me.
Dialplan :
exten = _*XXX***,n,ChanSpy(${SIPACC}) ; var $SIPACC has SIP peer
account name
Verbose :
[Jan 30 16:25:47] -- Executing [*204***@from-ITEL:10]
ChanSpy(SIP/itel0-2f21, itel1) in new stack
[Jan 30 16:25:48] -- SIP/itel0-2f21 Playing
'beep.alaw' (language 'nl')
But the spying IP-phone itel0 does not hear a thing. It should here
the conversation between SIP/itel1-2f10 and SIP/ITELin-2f0d
These are the 2 channels which are talking to each other :
SIP/itel1-2f10 SIP/ITELin-2f0d
Any idea which setting I'm missing ?
Kind regards,
Jonas.
On 01/25/2012 11:10 AM, Ishfaq Malik wrote:
I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio coming in and out of the channel
being spied on.'
Have you tried giving it a go?
-= Info about application 'ChanSpy' =-
[Synopsis]
Listen to a channel, and optionally whisper into it.
[Description]
This application is used to listen to the audio from an Asterisk
channel.
This includes the audio coming in and out of the channel being spied
on.
If the 'chanprefix' parameter is specified, only channels beginning with
this
string will be spied upon.
While spying, the following actions may be performed:
- Dialing '#' cycles the volume level.
- Dialing '*' will stop spying and look for another channel to spy on.
- Dialing a series of digits followed by '#' builds a channel name to
append
to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing
the
digits '1234#' while spying will begin spying on the channel
'Agent/1234'.
Note that this feature will be overridden if the 'd' option is used
NOTE: The X option supersedes the three features above in that if a
valid
single digit extension exists in the correct context ChanSpy will exit
to
it. This also disables choosing a channel based on 'chanprefix' and a
digit
sequence.
[Syntax]
ChanSpy([chanprefix][,options])
[Arguments]
options
b: Only spy on channels involved in a bridged call.
B: Instead of whispering on a single channel barge in on both
channels
involved in the call.
c(digit):
digit - Specify a DTMF digit that can be used to spy on the
next available channel.
d: Override the typical numeric DTMF functionality and instead use
DTMF to switch between spy modes.
4 - spy mode
5 - whisper mode
6 - barge mode
e(ext): Enable *enforced* mode, so the spying channel can only
monitor
extensions whose name is in the ext : delimited list.
E: Exit when the spied-on channel hangs up.
g(grp):
grp - Only spy on channels in which one or more of the groups
listed in grp matches one or more groups from the ${SPYGROUP}
variable set on the channel to be spied upon.
NOTE: both grp and ${SPYGROUP} can contain either a single group
or a colon-delimited list of groups, such as
'sales:support:accountin
g'.
n([mailbox][@context]): Say the name of the person being spied on
if that person has recorded his/her name. If a context is specified,
then
that voicemail context will be searched when retrieving the name,
otherwise
the 'default' context be used when searching for the name (i.e. if
SIP/1000
is the channel being spied on and no mailbox is specified, then
'1000'
will be used when searching for the name).
o: Only listen to audio coming from this channel.
q: Don't play a beep when beginning to spy on a channel, or speak
the selected channel name.
r([basename]): Record the session to the monitor spool directory.
An optional base for the filename may be specified. The default is
'
chanspy'.
s: Skip the playback of the channel type (i.e. SIP, IAX, etc) when
speaking the selected channel name.
S: Stop when no more channels are left to spy on.
v([value]): Adjust the initial volume in the range from '-4' to
'4'. A negative value refers to a quieter setting.
w: Enable 'whisper' mode, so the spying channel can talk to the
spied-on channel.
W: Enable 'private whisper' mode, so the spying channel can talk
to the spied-on channel but cannot listen to that channel.
x(digit):
digit - Specify a DTMF digit that can be used to exit the
application.
X: Allow the user to exit ChanSpy to a valid single digit numeric
extension in the current context or the context
Re: [asterisk-users] Asterisk 1.8.9.0 Now Available
We mirror off http://packages.asterisk.org to a staging server, then update from there. When will this show up on packages.asterisk.org? Thanks! EKG -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.9.0 Now Available
On 01/30/2012 11:06 AM, Eric Germann wrote: We mirror off http://packages.asterisk.org to a staging server, then update from there. When will this show up on packages.asterisk.org? Thanks! EKG The RPMs should be there in a few minutes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec
Anyone using the G729 codec to create a h.323 trunk between an Avaya Communication manager and Asterisk Freepbx System and works? I don't have the G729 codec installed on the Asterisk and running G711MU on avaya and getting invalid codec when calling from Avaya to Asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.9.0 Now Available
Thanks! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Monday, January 30, 2012 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8.9.0 Now Available On 01/30/2012 11:06 AM, Eric Germann wrote: We mirror off http://packages.asterisk.org to a staging server, then update from there. When will this show up on packages.asterisk.org? Thanks! EKG The RPMs should be there in a few minutes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fall back to inband DTMF?
I have an issue where one of the carriers that my up-line is using is not offering RFC-2833. I am getting the response from them that if RFC-2833 or SIP INFO is not offered then I should fall back to inband. I only have RFC-2833 offered enabled on all phone sets and trunks. The Peer accounts are forcing RFC-2833. So how would I accomidate this issue as I never want inband used if RFC-2833 is offerd. As I understand it asterisk only has the option of one of the DTMF methods. Any ideas would be apperciated. Is there a way to not offer inband and still force asterisk to fall back and use it the carrier end forces it? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RFC 5922 (TLS Certificates) and Asterisk
I've raised a bug report about this here: https://issues.asterisk.org/jira/browse/ASTERISK-19268 I'm just wondering who else has been investigating RFC 5922 style certificate practices? Which CAs have been able to provide appropriate certificates? What kind of interoperability testing has been done between the major products (e.g. Asterisk, Kamailio, OpenSIPS, reSIProcate/repro)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TLS problems - patch in Jira
I've just come across this issue: https://issues.asterisk.org/jira/browse/ASTERISK-17727 I am strongly in support of TLS and I believe this issue will be a stumbling block for more and more users - because more and more CAs are using the intermediate certificate chains For example, the free startssl.com certs are trusted by Android phones now. I have a UA running on my phone against a SIP proxy with Kamailio. I have the free cert and the intermediate cert in a single pem file. It all works. As noted in the bug, there may be phones that don't supported chain certs - but that shouldn't prevent the rest of us using them. People with such phones (which are becoming the minority) can just not use chained certs. There is no reason not to apply the supplied patch - that patch for Asterisk just makes it use the same OpenSSL function that Kamailio is using to load the chain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CA Issued Certificates / TLS + SRTP
On 30/01/12 17:12, Stuart Elvish wrote: Hi all, Firstly, apologies if the answer to this question should be obvious. I have just started working with SRTP and had a self-signed certificate working perfectly. I have now purchased a CA signed certificate but can't get it to work properly with Asterisk. I think I have a configuration error. No, you've found a bug - I just posted an update about this issue yesterday, predicting people would get stuck on this issue: http://lists.digium.com/pipermail/asterisk-users/2012-January/269856.html The certificate is a GeoTrust Rapid SSL certificate. I have received the my server specific crt file and also an intermediate certificate. Intermediate certificates work for some user agents (e.g. my Polycom). There has been speculation that they won't work with some older UAs Ultimately, most of the budget priced certificates are signed with an intermediate cert, and OpenSSL supports it, so there is no reason Asterisk shouldn't support this. I am not sure of the following and would greatly appreciate if someone could give me some guidance: * Can I specify the intermediate and .crt files separately in the sip.conf file? (I am thinking of a process similar to Apache where you specify three different files; server specific certificate, chain file and key file.) No, for OpenSSL-based code (such as Asterisk), it works like this: http://lists.sip-router.org/pipermail/sr-users/2012-January/071771.html However, Asterisk needs to be patched first, as in bug 17727 * Should the intermediate and server specific certificates be combined into one certificate file? Yes, in the correct order Currently, Asterisk expects the key and cert together in the same file: I think that is bad, but that is the way it is: https://issues.asterisk.org/jira/browse/ASTERISK-19267 * And, is it necessary to use both my server specific certificate and the intermediate certificate on the telephones or will the telephones only require the server specific certificate? The phones should already have the root certificate for Geotrust, you should not deploy intermediate roots into the phones if you can avoid it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fall back to inband DTMF?
I went through the source code and now understand better how dtmfmode=auto works. In testing I was able to resolve this by setting dtmfmode=auto. After further testing I will deploy it to production and see if it breaks anything but I am hoping this will be resolved for the long term. Thanks Bryant From: Bryant Zimmerman brya...@zktech.com Sent: Monday, January 30, 2012 3:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] fall back to inband DTMF? I have an issue where one of the carriers that my up-line is using is not offering RFC-2833. I am getting the response from them that if RFC-2833 or SIP INFO is not offered then I should fall back to inband. I only have RFC-2833 offered enabled on all phone sets and trunks. The Peer accounts are forcing RFC-2833. So how would I accomidate this issue as I never want inband used if RFC-2833 is offerd. As I understand it asterisk only has the option of one of the DTMF methods. Any ideas would be apperciated. Is there a way to not offer inband and still force asterisk to fall back and use it the carrier end forces it? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found
On Mon, Jan 30, 2012 at 7:31 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 01/28/2012 10:22 AM, Din Assegaf wrote: The error message is misleading; you are having this problem because the 'm' line in the SDP with the 'audio' offer has a port number of 0 (zero)., which means it is not an active media stream offer. It does not make any sense for the SDP in an INVITE for a new call to have an m-line with a port number of zero. I'll improve the error message so that this sort of situation won't be as confusing in the Thank you Kevin sir, I will check my sip client. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with DTMF in Voicemail main
Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk. On 10.1.0 and trunk, I can't successfully enter the password for any mailbox in voicemailmain on my Aastra 480i phones. All four version work with a Snom cordless SIP phone. In 10.0.0 and 10.0.1 the Aastra works perfectly. So needless to say I'm back to running 10.0.1. The WAF is very low for stuff like that. I notice that comedian mail has instead of [] brackets. Does that mean it's on its way to being deprecated? Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
