Re: [asterisk-users] Microsoft CRM Integration
On 16 Jul 2013, at 04:10, Klaverstyn, David C wrote: I’m hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Hi, You've not said what you're trying to integrate... Creating tasks for calls, contact lookups, automatic case creation. Either way, all possible with ODBC and FreeTDS. Steve-- _ -- 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] Microsoft CRM Integration
On Tuesday 16 July 2013, Klaverstyn, David C wrote: Hi All, I'm hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Well, that's a bit of a vague request. If you just want to add the facility to click on a phone number in some HTML and set up a call, you can easily do that with no more than a simple CGI script. What sort of integration were you hoping to add? I'm not familiar with Microsoft anything; but if their CRM product has an API for writing plugins, then it should be possible to add Asterisk integration. Of course, being Microsoft, they may well -not- support user-written plugins (or pretend to, but deliberately make it hard for you to persuade them to work) precisely in order to force you to pay for their proprietary extensions. -- AJS Answers come *after* questions. -- _ -- 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] Extra Sound Packages
Maybe this is a stupid question. Are the files in Extra Sound Packages related to any product or are they just supplemental material? I searched the source files for some of the file names and didn't find any reference. jg -- _ -- 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] ignore 183 session progress in parallel call scenarios
I am sure I submitted the following alternative behaviour to the
bug-tracker in the past, but cannot find any reference to it. Here is the
patch I use to IMHO improve this behaviour.
In case it is not officially uploaded, I will state here that this code is
disclaimed and unencumbered as if uploaded to JIRA.
Regards,
Steve
On 15 July 2013 17:20, Hristo Trendev dist.li...@gmail.com wrote:
I think I have found the answer to my questions in the source code of Dial:
case AST_CONTROL_PROGRESS:
ast_verb(3, %s is making progress passing it to %s\n,
ast_channel_name(c), ast_channel_name(in));
/* Setup early media if appropriate */
if (single !caller_entertained
CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
if (single || (!single !pa-sentringing)) {
ast_indicate(in, AST_CONTROL_PROGRESS);
}
}
.
.
Asterisk will attempt to bridge the media only for the case of a single
outgoing channel, but at the same time it will happily forward progress
messages for parallel calls: (!single !pa-sentringing) as long as no 180
Ringing message was sent out to the caller yet. The questions still remains
if this should be reported as bug or if there is indeed a use case when
sending 183 progress message, without actually bridging the media stream is
desired.
On Mon, Jul 15, 2013 at 4:14 PM, Hristo Trendev dist.li...@gmail.comwrote:
Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means any progress announcements, including remotely generated ringback,
are not passed back to the caller.
This behavior is completely correct, because there is no way to know
which early media audio stream to pass back to the caller in a parallel
call scenario (as in this case several endpoint may indicate session
progress all at the same time).
The question is why is asterisk still sending 183 session progress back
to the caller if no audio is to be bridged before the 200 OK anyway? If 183
are not passed back to the caller, then at least a 180 Ringing that may
come from another endpoint will cause the calling endpoint to generate
local ringback. This won't happen if the caller has received a 183 already.
So it's a bit of a race condition as well - if the first endpoint to
reply sends a 183 session progress this means the caller will not hear
any ringback even if some of the other endpoints are sending back 180
Ringing.
The question is can I somehow block 183 messages from being passed back
to the calling endpoint when dialing several destinations in parallel? I
don't see a point (please correct me if I'm wrong) to pass only the 183 SIP
message back to the caller without the corresponding RTP stream, so it may
be much better to actually ignore it when dealing with parallel call
scenarios (bug?).
BR,
Hristo
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multiple_early_media
Description: Binary data
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Re: [asterisk-users] Microsoft CRM Integration
http://www.camrivox.com/products/flexor-cti-dynamics-crm/ -- Ringfree Communications David Wessell 828-575-0030 x101 From: Steven Howes steve-li...@geekinter.netmailto:steve-li...@geekinter.net Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Date: Tuesday, July 16, 2013 4:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Microsoft CRM Integration On 16 Jul 2013, at 04:10, Klaverstyn, David C wrote: I'm hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Hi, You've not said what you're trying to integrate... Creating tasks for calls, contact lookups, automatic case creation. Either way, all possible with ODBC and FreeTDS. Steve -- _ -- 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] Voice analytics
Does anyone know of a realtime voice analytic engine that works with asterisk 11+ ? We want to be able to listen on the conversation for key words in order to ensure compliance . The plan is to show these keywords onscreen, and remove them once the agent has covered the compliance issues. This would necessitate that the conversation is monitored and analysed in realtime as we can't do it post-call ;) Thanks Julian -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] FLAC script to convert from wav to FLAC and also with other 3 to 4 formats
Hi,
Below link is the script which i found while surfing, this script basically
converts your voice file to flac format, where the file is reduced to 50%.
http://legroom.net/files/software/convtoflac.sh
The quality is really good, I tested. this...
In large production environment this script can be used, only challenging
part, please make sure the CPU usage is within the limit while conversion.
Can be used like this,
exten =
_4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,b,/root/
flac.sh ${MIXMONITOR_FILENAME}.wav)
Regards,
Gopal.
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Re: [asterisk-users] ignore 183 session progress in parallel call scenarios
Thanks Steve!
I too believe that this is indeed much better handling of 183 replies in a
parallel call. After testing for several hours today I actually wen't a bit
further (see ASTERISK-22082) and proposing to ignore the 183 altogether as
far as parallel calls are concerned. In my case I don't even need to
convert the 183 to 180, because all upstream providers are sending a 180 in
addition to the 183 and as long no 183 was sent to the caller prior to that
it just works...well almost always, see note 1 in the ticket above.
At first I was also tempted to simply convert it from 183 to 180, but then
I figured out that the 183 may sometimes contain an announcement as opposed
to a ringback tone. The caller won't hear it anyway, but simply sending 180
in this case will result in ringback being generated too early and not when
a real 180 ringing (possibly from another call leg) is received.
Ideally this can be a per sip peer or dialplan option, because some
providers will only send 183 and no 180 and in this case your patch will
work great, while others may send both, so simply ignoring the 183 will
provide ringback only when there is indeed a ringback tone and not some
other announcement (for example subscriber not available or similar), but
that's probably pushing it to far ;)
Whatever the solution, I think it will certainly be better than providing
no ringback at all.
Best,
Hristo
On Tue, Jul 16, 2013 at 12:33 PM, Steve Davies davies...@gmail.com wrote:
I am sure I submitted the following alternative behaviour to the
bug-tracker in the past, but cannot find any reference to it. Here is the
patch I use to IMHO improve this behaviour.
In case it is not officially uploaded, I will state here that this code is
disclaimed and unencumbered as if uploaded to JIRA.
Regards,
Steve
On 15 July 2013 17:20, Hristo Trendev dist.li...@gmail.com wrote:
I think I have found the answer to my questions in the source code of
Dial:
case AST_CONTROL_PROGRESS:
ast_verb(3, %s is making progress passing it to %s\n,
ast_channel_name(c), ast_channel_name(in));
/* Setup early media if appropriate */
if (single !caller_entertained
CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
if (single || (!single !pa-sentringing)) {
ast_indicate(in, AST_CONTROL_PROGRESS);
}
}
.
.
Asterisk will attempt to bridge the media only for the case of a single
outgoing channel, but at the same time it will happily forward progress
messages for parallel calls: (!single !pa-sentringing) as long as no 180
Ringing message was sent out to the caller yet. The questions still remains
if this should be reported as bug or if there is indeed a use case when
sending 183 progress message, without actually bridging the media stream is
desired.
On Mon, Jul 15, 2013 at 4:14 PM, Hristo Trendev dist.li...@gmail.comwrote:
Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means any progress announcements, including remotely generated ringback,
are not passed back to the caller.
This behavior is completely correct, because there is no way to know
which early media audio stream to pass back to the caller in a parallel
call scenario (as in this case several endpoint may indicate session
progress all at the same time).
The question is why is asterisk still sending 183 session progress back
to the caller if no audio is to be bridged before the 200 OK anyway? If 183
are not passed back to the caller, then at least a 180 Ringing that may
come from another endpoint will cause the calling endpoint to generate
local ringback. This won't happen if the caller has received a 183 already.
So it's a bit of a race condition as well - if the first endpoint to
reply sends a 183 session progress this means the caller will not hear
any ringback even if some of the other endpoints are sending back 180
Ringing.
The question is can I somehow block 183 messages from being passed back
to the calling endpoint when dialing several destinations in parallel? I
don't see a point (please correct me if I'm wrong) to pass only the 183 SIP
message back to the caller without the corresponding RTP stream, so it may
be much better to actually ignore it when dealing with parallel call
scenarios (bug?).
BR,
Hristo
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[asterisk-users] Help with decyphering DND status
Arch x86_64 OS CentOS-6.4 (freepbx) Asterisk 11.4 FreePBX 2.11.0.4 Snom870 with FW-8.7.4.8 What I am attempting to do is to set a different background colour for the BLF vkeys when a station is set to DND. This is supposedly accomplished through this setting in the phones provisioning file: vkey_blue perm=RW DND Blue.on Blue.pickup Blue.park Blue.message /vkey_blue However, this does not work. What instead works when DND is set is this: vkey_blue perm=RW CONNECTED Blue.on Blue.pickup Blue.park Blue.message /vkey_blue Which makes no sense to me. However, I infer that somewhere in the bowels of Asterisk something is set for DND which the Snom interprets as CONNECTED instead. It is what this something is and how it is set that I am trying to understand. To further this I am trying to discover is exactly what is sent to the other stations by asterisk when DND is enabled for a station. Short of installing wireshark is there any other way to see exactly what asterisk is sending to the phone? When I look at the SIP trace logs on the handset when switch dnd on and off on another handset I see this sort of thing: Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:14:422 (693 bytes): NOTIFY sip:41720@192.168.6.120:3072;line=d241fk25 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK56d306e4;rport Max-Forwards: 70 From: sip:41712@192.168.6.9;;tag=as149ada79 To: sip:41720@192.168.6.9;tag=tyybvtkyiy Contact: sip:41712@192.168.6.9:5060 Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i CSeq: 191 NOTIFY User-Agent: FPBX-2.11.0(11.4.0) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 206 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=89 state=full entity=sip:41712@192.168.6.9 dialog id=41712 stateconfirmed/state /dialog /dialog-info Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:14:426 (300 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK56d306e4;rport=5060 From: sip:41712@192.168.6.9;;tag=as149ada79 To: sip:41720@192.168.6.9;tag=tyybvtkyiy Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i CSeq: 191 NOTIFY User-Agent: snom870/8.7.4.8 Content-Length: 0 Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:16:051 (693 bytes): NOTIFY sip:41720@192.168.6.120:3072;line=d241fk25 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK49b47181;rport Max-Forwards: 70 From: sip:41712@192.168.6.9;;tag=as149ada79 To: sip:41720@192.168.6.9;tag=tyybvtkyiy Contact: sip:41712@192.168.6.9:5060 Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i CSeq: 192 NOTIFY User-Agent: FPBX-2.11.0(11.4.0) Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 206 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=90 state=full entity=sip:41712@192.168.6.9 dialog id=41712 stateconfirmed/state /dialog /dialog-info Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:16:055 (300 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.6.9:5060;branch=z9hG4bK49b47181;rport=5060 From: sip:41712@192.168.6.9;;tag=as149ada79 To: sip:41720@192.168.6.9;tag=tyybvtkyiy Call-ID: 455FE451B390010B19F8CAE0BE485B25-ju0bdbwv885i CSeq: 192 NOTIFY User-Agent: snom870/8.7.4.8 Content-Length: 0 Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:16:672 (483 bytes): SUBSCRIBE sip:41710@192.168.6.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.120:3072;branch=z9hG4bK-f20yldu080fk;rport From: sip:41720@192.168.6.9;tag=se3w15c5fb To: sip:41710@192.168.6.9;;tag=as6723ebb5 Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz CSeq: 63 SUBSCRIBE Max-Forwards: 70 User-Agent: snom870/8.7.4.8 Contact: sip:41720@192.168.6.120:3072;line=d241fk25;reg-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 Received from udp:192.168.6.9:5060 at 16/7/2013 11:56:16:674 (529 bytes): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.6.120:3072;branch=z9hG4bK-f20yldu080fk;received=192.168.6.120;rport=3072 From: sip:41720@192.168.6.9;tag=se3w15c5fb To: sip:41710@192.168.6.9;;tag=as6723ebb5 Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz CSeq: 63 SUBSCRIBE Server: FPBX-2.11.0(11.4.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=510db654 Content-Length: 0 Sent to udp:192.168.6.9:5060 at 16/7/2013 11:56:16:680 (648 bytes): SUBSCRIBE sip:41710@192.168.6.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.120:3072;branch=z9hG4bK-6lft658u13gn;rport From: sip:41720@192.168.6.9;tag=se3w15c5fb To: sip:41710@192.168.6.9;;tag=as6723ebb5 Call-ID: 455FE4519A6B01B916F8CAE0BE485B25-x08c2euu13zz CSeq: 64 SUBSCRIBE Max-Forwards: 70 User-Agent: snom870/8.7.4.8 Contact: sip:41720@192.168.6.120:3072;line=d241fk25;reg-id=1 Event: dialog Accept: application/dialog-info+xml Authorization: Digest
Re: [asterisk-users] Mailing a fax with mutt does not succeed
Hello everyone, I'd changed the server and mutt started working, but I'll
test your advices and wil let you lnow ass soon as I can.
Thank you!
Elder
On Mon, Jun 24, 2013 at 7:38 AM, Larry Moore lmo...@omninet.net.au wrote:
On 22/06/2013 2:17 PM, Steve Edwards wrote:
On Sat, 22 Jun 2013, Larry Moore wrote:
echo $MSGFILE
printf %18s Sender: $MSGFILE; printf %-20s\n
${REMOTESTATIONID} $MSGFILE
printf %18s Pages: $MSGFILE; printf %-20s\n ${FAXPAGES}
$MSGFILE
printf %18s Signal Rate: $MSGFILE; printf %-20s\n
${FAXBITRATE} bit/s $MSGFILE
printf %18s CallerID Number: $MSGFILE; printf %-20s\n
${CIDNUMBER} $MSGFILE
printf %18s CallerID Name: $MSGFILE; printf %-20s\n
${CIDNAME} $MSGFILE
printf %18s Call Duration: $MSGFILE; printf %-20s\n
${DURATION} $MSGFILE
printf %18s Status: $MSGFILE; printf %-20s\n ${FAXERROR}
$MSGFILE
How about:
#!/bin/bash
FORMAT='%18s %-20s\n'
(
printf ${FORMAT} 'Sender:'${REMOTESTATIONID}
printf ${FORMAT} 'Pages:'${FAXPAGES}
printf ${FORMAT} 'Signal Rate:'${FAXBITRATE} bits/s
printf ${FORMAT} 'CallerID Number:'${CIDNUMBER}
printf ${FORMAT} 'CallerID Name:'${CIDNAME}
printf ${FORMAT} 'Call Duration:'${DURATION}
printf ${FORMAT} 'Status:'${FAXERROR}
) ${MSGFILE}
Thank Steve,
That makes the section more readable and it works with /bin/ksh too.
Cheers,
Larry.
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Re: [asterisk-users] PoE module
Here's a cheap solution for PoE piggybacked over your existing network. http://www.amazon.com/gp/product/B0002R6X9S On Jul 14, 2013, at 3:12 PM, bilal ghayyad wrote: Hello; We have a cisco switches but they are not PoE and we need only to have PoE device so the cables come for it first to provide the power and then goes to the switch (to be like batch panel), is there something like this that can be used for the IP Phones? Regards Bilal -- This message has been scanned for viruses and dangerous content and is believed to be clean. -- _ -- 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
[asterisk-users] SIP timers
Hello List, I tried to change the following parameters in sip.conf file, but looks like it cannot be changed, Defaut values: ;t1min=100 ;timert1=500 ;timerb=32000 I have changed to: ;t1min=100 timert1=100 timerb=6400 Sometime I can see too many retransmission of BYE to some of the UAs if UA is unreachable. Is there a way that I can reduce the number of retransmission of BYE message? Regards Rajib -- _ -- 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
