Re: [asterisk-users] SSRC =0x0 in RTP
> > Hi Joshua, > Thank you for looking into this. > Their response IS based on traces I've sent them. Attached is such trace in text format (server IP has been changed to 111.111.111.111). Some repeating RTP packets has been truncated. You can see that after the 200 OK SSRC is sent from the server to the phone as '0x0'. The same has happened with G729 codec. > Let me know if you need the full trace or anything else from my side. > I should also mention that this is Asterisk version 1.8.12.1 > Thank you > Harel > -- > On Tue, Nov 14, 2017, at 09:14 AM, Harel Cohen wrote: > > Hello, > > I have a problem where on an outgoing call a Grandstream phone > > (GXP2130) closes the incoming voice stream about 1 second into the > > call (the remote party hears the Grandstream, the Grandstream doesn't > > hear thr remote party). I have verified with logs and traces that this > > is not a NAT issue or any other network-related problem. All incoming > > RTP packets arrive at the phone on the correct port etc. as declared in the SDP. > > I opened a ticket with Grandstream and they replied: " > > > > *the phone starts receiving RTP with SSRC =0x0 which is wrong".* > > > > Is this an Asterisk problem or the phones? Is this something that can > > be fixed on the Asterisk side? > Asterisk would be sending the RTP to the Grandstream. I'd suggest getting a packet capture using tcpdump or wireshark to confirm what they've said though. I just looked at the code and I don't see a way that we'd ever have the SSRC be 0. > Cheers, > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org > > No. Time SourceDestination Protocol Length Info 1 0.00 1224 Frame 1: 1224 bytes on wire (9792 bits), 1224 bytes captured (9792 bits) This frame is marked as ignored No. Time SourceDestination Protocol Length Info 2 0.078267 195.191.156.50192.168.10.144SIP 631 Status: 401 Unauthorized | Frame 2: 631 bytes on wire (5048 bits), 631 bytes captured (5048 bits) Ethernet II, Src: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c), Dst: Grandstr_a7:a0:fa (00:0b:82:a7:a0:fa) 802.1Q Virtual LAN, PRI: 0, DEI: 0, ID: 0 Internet Protocol Version 4, Src: 195.191.156.50, Dst: 192.168.10.144 User Datagram Protocol, Src Port: 55060, Dst Port: 21321 Session Initiation Protocol (401) No. Time SourceDestination Protocol Length Info 3 0.090271 192.168.10.144195.191.156.50SIP 431 Request: ACK sip:0035699143...@sip1.mayorcom.com:55060 | Frame 3: 431 bytes on wire (3448 bits), 431 bytes captured (3448 bits) Ethernet II, Src: Grandstr_a7:a0:fa (00:0b:82:a7:a0:fa), Dst: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c) Internet Protocol Version 4, Src: 192.168.10.144, Dst: 195.191.156.50 User Datagram Protocol, Src Port: 21321, Dst Port: 55060 Session Initiation Protocol (ACK) No. Time SourceDestination Protocol Length Info 4 0.120599 192.168.10.144195.191.156.50SIP/SDP 1427 Request: INVITE sip:0035699143...@sip1.mayorcom.com:55060 | Frame 4: 1427 bytes on wire (11416 bits), 1427 bytes captured (11416 bits) Ethernet II, Src: Grandstr_a7:a0:fa (00:0b:82:a7:a0:fa), Dst: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c) Internet Protocol Version 4, Src: 192.168.10.144, Dst: 195.191.156.50 User Datagram Protocol, Src Port: 21321, Dst Port: 55060 Session Initiation Protocol (INVITE) No. Time SourceDestination Protocol Length Info 5 0.200309 195.191.156.50192.168.10.144SIP 576 Status: 100 Trying | Frame 5: 576 bytes on wire (4608 bits), 576 bytes captured (4608 bits) Ethernet II, Src: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c), Dst: Grandstr_a7:a0:fa (00:0b:82:a7:a0:fa) 802.1Q Virtual LAN, PRI: 0, DEI: 0, ID: 0 Internet Protocol Version 4, Src: 195.191.156.50, Dst: 192.168.10.144 User Datagram Protocol, Src Port: 55060, Dst Port: 21321 Session Initiation Protocol (100) No. Time SourceDestination Protocol Length Info 6 1.742472 195.191.156.50192.168.10.144SIP/SDP 883 Status: 183 Session Progress | Frame 6: 883 bytes on wire (7064 bits), 883 bytes captured (7064 bits) Ethernet II, Src: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c), Dst: Grandstr_a7:a0:fa (00:0b:82:a7:a0:fa) 802.1Q Virtual LAN, PRI: 0, DEI: 0, ID: 0 Internet Protocol Version 4, Src: 195.191.156.50, Dst: 192.168.10.144 User Datagram Protocol, Src P
[asterisk-users] SSRC =0x0 in RTP
Hello, I have a problem where on an outgoing call a Grandstream phone (GXP2130) closes the incoming voice stream about 1 second into the call (the remote party hears the Grandstream, the Grandstream doesn't hear thr remote party). I have verified with logs and traces that this is not a NAT issue or any other network-related problem. All incoming RTP packets arrive at the phone on the correct port etc. as declared in the SDP. I opened a ticket with Grandstream and they replied: " *the phone starts receiving RTP with SSRC =0x0 which is wrong".* Is this an Asterisk problem or the phones? Is this something that can be fixed on the Asterisk side? Thank you, Harel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem using Android SIP-Client
Hi, Is the Sophos a home router or professional one? In many cases what home router does by default needs to be configured manually on professional one. E.G. a home router will allow outgoing sessions and create a return path by default where professional one won't. Two things I would look for: 1. Look for, and disable, ALG for SIP. The idea of ALG is nice but I haven't encountered a device that implements this properly (I'm not a network expert so it doesn't mean that there isn't such a router out there). 2. On the Sophos try to statically open the UDP port range of your RTP to outgoing traffic from your phone to your SIP server. Note that outgoing port range is what your SIP server defines as its port range, not your phone. If you get one way voice (remote hears phone) then you are on the right direction. You'll then need to open the incoming ports too for the ports that your phone is expecting to get its RTP from. KR Harel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor Chan State Change over AMI
Hello All, For a development of an auto-dialler I need to make a script that will SUBSCRIBE to the status of certain SIP channels and act upon the changes of their state. How can I subscribe over AMI (what 'Action:' do I need to use) and what response should I look for? Many Thanks, Harel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't install gmime22
No one could assist? Could someone please tell me on which repository I can find Gmime22-devel for 64-bit Centos6.5? Is gmime-devel good or do I need to have gmime22-devel? What will happen if I don't install gmime22? Thank you... Harel Message: 3 Date: Mon, 6 Jul 2015 02:53:51 +0200 From: Harel Cohen ha...@mayorcom.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't install gmime22 Message-ID: 00a601d0b786$3a7bffd0$af73ff70$@mayorcom.com Content-Type: text/plain; charset=us-ascii Hello list, I'm trying to install gmime22 package which is one of the packages reported as required by ./contrib/scripts/install_prereq test. Whatever I do I'm getting to a dead end. On the regular yum repositories that I use (centos, epel, rpmforge, asterisk, digium) it is not found. I've found it on Fedora repositories however trying to use those I get all sorts of errors: On fedora17 repository: ERROR You need to update rpm to handle: rpmlib(X-CheckUnifiedSystemdir) is needed by filesystem-3-2.fc17.x86_64 When I try to update rpm I'm getting a conflict between some systemd package to kernel On fedora21 repository: Not found On fedora20 repository it is reported as installed but with these errors: Error unpacking rpm package filesystem-3.2-19.fc20.x86_64 error: unpacking of archive failed on file /bin: cpio: rename Cleanup: glibc-common-2.12-1.149.el6_6.9.x86_64 7/10 Cleanup: bash-4.1.2-29.el6.x86_64 8/10 Non-fatal POSTUN scriptlet failure in rpm package bash warning: %postun(bash-4.1.2-29.el6.x86_64) scriptlet failed, exit status 127 Cleanup: glibc-2.12-1.149.el6_6.9.x86_64 9/10 warning: /etc/localtime saved as /etc/localtime.rpmsave ...and also the system hang on shutdown and won't boot again Could you please advise how to properly install this package? I'm on CentOS 6.5 with updates, running on AMD Athlon +4400 and 64bit Thank you... -- _ -- 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] Can't install gmime22
Hello list, I'm trying to install gmime22 package which is one of the packages reported as required by ./contrib/scripts/install_prereq test. Whatever I do I'm getting to a dead end. On the regular yum repositories that I use (centos, epel, rpmforge, asterisk, digium) it is not found. I've found it on Fedora repositories however trying to use those I get all sorts of errors: On fedora17 repository: ERROR You need to update rpm to handle: rpmlib(X-CheckUnifiedSystemdir) is needed by filesystem-3-2.fc17.x86_64 When I try to update rpm I'm getting a conflict between some systemd package to kernel On fedora21 repository: Not found On fedora20 repository it is reported as installed but with these errors: Error unpacking rpm package filesystem-3.2-19.fc20.x86_64 error: unpacking of archive failed on file /bin: cpio: rename Cleanup: glibc-common-2.12-1.149.el6_6.9.x86_64 7/10 Cleanup: bash-4.1.2-29.el6.x86_64 8/10 Non-fatal POSTUN scriptlet failure in rpm package bash warning: %postun(bash-4.1.2-29.el6.x86_64) scriptlet failed, exit status 127 Cleanup: glibc-2.12-1.149.el6_6.9.x86_64 9/10 warning: /etc/localtime saved as /etc/localtime.rpmsave ...and also the system hang on shutdown and won't boot again Could you please advise how to properly install this package? I'm on CentOS 6.5 with updates, running on AMD Athlon +4400 and 64bit Thank you... -- _ -- 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] RTP sent to remote internal IP
Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the public IP address as it is seen on the packet header. Signalling is flowing correctly with no issues. Could you please advise why is this happening and how to correct this? Here is the [peer] in my sip.conf and the SDP in the setup (INVITE + OK). I'll be happy to provide any other information if needed: Sip.conf: [peer_name] deny=0.0.0.0/0 permit=remote_public_IP type=peer host=remote_public_IP ; same as permit defaultip=remote_public_IP ; same as permit qualify=no nat=yes disallow=all allow=alaw context=CALL_in dtmfmode=rfc2833 codecprobe=yes canreinvite=yes video=no restrictcid=no insecure=invite trustrpid = yes Below is the SDP from both the INVITE and OK packets (from TShark). 172.24.100.2 is the local-private IP address of the remote UA and 192.168.1.200 is the local-private IP of my Asterisk. Both public IP's are static and do not change: *INVITE* Internet Protocol Version 4, Src: 192.168.1.200 (192.168.1.200), Dst: remote_public_IP (remote_public_IP) User Datagram Protocol, Src Port: 65060 (65060), Dst Port: 5060 (5060) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:858@remote_public_IP SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 614275163 614275163 IN IP4 my_public_IP Session Name (s): Quantum PBX V 3.0 1.8.17.1 cert Connection Information (c): IN IP4 my_public_IP Time Description, active time (t): 0 0 Media Description, name and address (m): audio 18468 RTP/AVP 8 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ptime:20 Media Attribute (a): sendrecv *200 OK* Internet Protocol Version 4, Src: remote_public_IP (remote_public_IP), Dst: 192.168.1.200 (192.168.1.200) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 65060 (65060) Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): default 1426152411 1426152411 IN IP4 172.24.100.2 Session Name (s): Quantum PBX V 3.0 1.8.17.1 cert Connection Information (c): IN IP4 172.24.100.2 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 32000 RTP/AVP 8 101 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): silenceSupp:off - - - - Media Attribute (a): ptime:20 Media Attribute (a): maxptime:90 After this 'OK' RTP packets are sent to 172.24.100.2 instead of remote_public_IP Thank you, Harel -- _ -- 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] RTP sent to internal IP
Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the public IP address as it is seen on the packet header. Signalling is flowing correctly with no issues. Could you please advise why is this happening and how to correct this? Here is the [peer] in my sip.conf and the SDP in the setup (INVITE + OK). I'll be happy to provide any other information if needed: Sip.conf: [peer_name] deny=0.0.0.0/0 permit=remote_public_IP type=peer host=remote_public_IP ; same as permit defaultip=remote_public_IP ; same as permit qualify=no nat=yes disallow=all allow=alaw context=CALL_in dtmfmode=rfc2833 codecprobe=yes canreinvite=yes video=no restrictcid=no insecure=invite trustrpid = yes The SDP from both the INVITE and OK packets (from TShark). 172.24.100.2 is the local-private IP address of the remote UA and 192.168.1.200 is the local-private IP of my Asterisk. Both public IP's are static and do not change: *INVITE* Internet Protocol Version 4, Src: 192.168.1.200 (192.168.1.200), Dst: remote_public_IP (remote_public_IP) User Datagram Protocol, Src Port: 65060 (65060), Dst Port: 5060 (5060) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:858@remote_public_IP SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 614275163 614275163 IN IP4 my_public_IP Session Name (s): Quantum PBX V 3.0 1.8.17.1 cert Connection Information (c): IN IP4 my_public_IP Time Description, active time (t): 0 0 Media Description, name and address (m): audio 18468 RTP/AVP 8 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ptime:20 Media Attribute (a): sendrecv *200 OK* Internet Protocol Version 4, Src: remote_public_IP (remote_public_IP), Dst: 192.168.1.200 (192.168.1.200) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 65060 (65060) Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): default 1426152411 1426152411 IN IP4 172.24.100.2 Session Name (s): Quantum PBX V 3.0 1.8.17.1 cert Connection Information (c): IN IP4 172.24.100.2 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 32000 RTP/AVP 8 101 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): silenceSupp:off - - - - Media Attribute (a): ptime:20 Media Attribute (a): maxptime:90 Thank you, Harel -- _ -- 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] Populate CDR issues
Hi Mike,
I've tried updating my CDR's via the h exten but with no success. I've tried
with both endbeforehexten=no and endbeforehexten=yes (in cdr.conf) but the
value refused to appear in my CDR (even though I see the Set() application
being executed in the console under the h exten). Thank you for your suggestion
though...
Any other thoughts are welcome.
Kind Regards,
Harel Cohen
-Original Message-
Date: Mon, 12 Dec 2011 13:41:31 -0700
From: Mike Diehl mdi...@diehlnet.com
Subject: Re: [asterisk-users] Populate CDR issues
To: asterisk-users@lists.digium.com
Cc: Harel Cohen ha...@easycall.gi
Message-ID: 201112121341.32142.mdi...@diehlnet.com
Content-Type: Text/Plain; charset=iso-8859-1
On Monday 12 December 2011 4:28:17 am Harel Cohen wrote:
Danny,
Why would you think this is a circumvent? I'm using a nice feature
of 1.8 where I can create any CDR field I like and populate it by
using the
CDR(fieldname) function. While all other fields that I created are
populated properly (however before the 'dial' commences) it seems like
at this point of the dial plan the CDR is closed for editing even
though I configured endbeforehexten=no in my cdr.conf.
I agree, this is a perfectly valid use of the CDR. I do the same thing, btw.
I think what you are seeing is that when your call starts, Asterisk creates a
record, either in memory, or in a db transaction. When the call is torn down,
the record is updated and committed to the db. The down-shot is that any
changes you make to the db record get clobbered by this last update.
I ended up making some of my updates in the hang-up phase via the h
extension. See if that will do what you need.
--
Take care and have fun,
Mike Diehl.
--
Date: Thu, 1 Dec 2011 12:57:56 +0100
From: Harel Cohen ha...@easycall.gi
Subject: [asterisk-users] Populate CDR issues
To: asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Message-ID:
CABC006234837141A279831F68041406CABE9415A4@ECS.EasyCall.local
Content-Type: text/plain; charset=us-ascii
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel. I've tried using the option
'U' with my dial command (execute subroutine for called channel after called
channel answered but before the call is bridged). While this throws the correct
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC
and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application where
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015, SIP/office_ServerRoom,,FgU(jump2SetVar))
in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)})
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016, CDR(chanoutsigip)=192.168.20.226:5065)
in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
Re: [asterisk-users] Populate CDR issues
Danny,
Why would you think this is a circumvent? I'm using a nice feature of 1.8
where I can create any CDR field I like and populate it by using the
CDR(fieldname) function. While all other fields that I created are populated
properly (however before the 'dial' commences) it seems like at this point of
the dial plan the CDR is closed for editing even though I configured
endbeforehexten=no in my cdr.conf.
It might be related to issue ASTERISK-18875 as suggested by Daniel (Vol. 89
issue 8, topic 9).
I'll be happy to know if someone has a different knowledge on the subject,
otherwise I'll simply follow ASTERISK-18875. My problem with this issue is that
it is defined as low importance which means that it will probably take long to
handle if at all...
Harel
**
Message: 4
Date: Tue, 6 Dec 2011 07:29:54 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Populate CDR issues
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 009801ccb41b$24417cd0$6cc47670$@debsinc.com
Content-Type: text/plain; charset=us-ascii
IMO you are trying to circumvent basic Asterisk functionality. It's your CDR
so you can do what you want with it - I think the answer to this is to populate
another DB with the live call data, then update the CDR from that after the
call has ended (perhaps a daemon).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harel Cohen
Sent: Tuesday, December 06, 2011 3:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Populate CDR issues
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just in case
someone who might be able to help missed my previous post.
Thank You.
*
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel. I've tried using the option
'U' with my dial command (execute subroutine for called channel after called
channel answered but before the call is bridged). While this throws the correct
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC
and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application where
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015,
SIP/office_ServerRoom,,FgU(jump2SetVar)) in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peer
name)},port)}) ; resulting format: a.b.c.d:port
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutsigip)=192.168.20.226:5065) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Auto fallthrough, channel
'SIP/office_ServerRoom-0016
[asterisk-users] Populate CDR issues
Hello Everyone,
I didn't get a reply to my problem below so I'm posting again just in case
someone who might be able to help missed my previous post.
Thank You...
*
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel. I've tried using the option
'U' with my dial command (execute subroutine for called channel after called
channel answered but before the call is bridged). While this throws the correct
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC
and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application where
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015, SIP/office_ServerRoom,,FgU(jump2SetVar))
in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)})
; resulting format: a.b.c.d:port
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016, CDR(chanoutsigip)=192.168.20.226:5065)
in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Auto fallthrough, channel
'SIP/office_ServerRoom-0016' status is 'UNKNOWN'
[Dec 1 12:14:33] -- Remotely bridging SIP/office_Admin2-0015 and
SIP/office_ServerRoom-0016
When call is terminated the relevant fields in the database for
CDR(chanoutsigip), CDR(chanoutmediaip) and CDR(chanoutcodec) are populated with
their default values (typically blank or '-') and NOT with the values above.
Am I doing something wrong or is there a different way to populate CDR's with
info from called channel (leg B)?
Thank you for your replies...
Harel
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[asterisk-users] Populate CDR issues
Hello list,
I'm trying to populate my CDR logs with values which are available after the
call has started (e.g. signalling IP of remote user, media IP, codec etc.).
While CHANNEL function give me all I need for the incoming leg (leg A), I can't
get the relevant values for the outgoing channel. I've tried using the option
'U' with my dial command (execute subroutine for called channel after called
channel answered but before the call is bridged). While this throws the correct
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC
and the table therein contains the relevant fields.
This is the console with 'very-verbose' output for the 'Dial' application where
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP
192.168.20.226. My comments added prefixed by ** and on separate line:
** channel here is source channel: SIP/office_Admin2-0015
[Dec 1 12:14:31] -- Executing [316@InternalDP:5]
Dial(SIP/office_Admin2-0015, SIP/office_ServerRoom,,FgU(jump2SetVar))
in new stack
[Dec 1 12:14:31] == Using UDPTL CoS mark 5
[Dec 1 12:14:31] == Using SIP RTP CoS mark 5
[Dec 1 12:14:31] -- Called SIP/office_ServerRoom
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec 1 12:14:33] -- SIP/office_ServerRoom-0016 answered
SIP/office_Admin2-0015
** from here the channel is the destination channel:
SIP/office_ServerRoom-0016
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:1]
Gosub(SIP/office_ServerRoom-0016, SetVar,postdial,1) in new stack
** This is how I obtain channel information:
** exten =
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)})
** same = n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same = n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial@SetVar:1]
Set(SIP/office_ServerRoom-0016, CDR(chanoutsigip)=192.168.20.226:5065)
in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:2]
Set(SIP/office_ServerRoom-0016,
CDR(chanoutmediaip)=192.168.20.226:23008) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:3]
Set(SIP/office_ServerRoom-0016, CDR(chanoutcodec)=g729) in new stack
[Dec 1 12:14:33] -- Executing [postdial@SetVar:4]
Goto(SIP/office_ServerRoom-0016, endsub,1) in new stack
[Dec 1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec 1 12:14:33] -- Executing [endsub@SetVar:1]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@jump2SetVar:2]
Return(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1]
NoOp(SIP/office_ServerRoom-0016, ) in new stack
[Dec 1 12:14:33] -- Auto fallthrough, channel
'SIP/office_ServerRoom-0016' status is 'UNKNOWN'
[Dec 1 12:14:33] -- Remotely bridging SIP/office_Admin2-0015 and
SIP/office_ServerRoom-0016
When call is terminated the relevant fields in the database for
CDR(chanoutsigip), CDR(chanoutmediaip) and CDR(chanoutcodec) are populated with
their default values (typically blank or '-') and NOT with the values above.
Am I doing something wrong or is there a different way to populate CDR's with
info from called channel (leg B)?
Thank you for your replies...
Harel
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[asterisk-users] Play different voice-mail messages based on certain conditions
Hello List, I have few installations out there based on 1.6.1 or above. I'm trying to play different voice mail messages based on certain criteria's. For example, I want during office hours to play (in short): we are not available to take your call, please leave a message, during off-hours and weekends I would play: we are closed, our opening hours xx:xx-yy:yy, please leave a message or send a fax or send an email and during holidays I would play: we are closed due to holiday, please leave a message, fax, blab la etc. I've tried to configure context for each case and set the directoryintro in each such context however the Asterisk was always looking for vm-intro and it was always looking for it in /var/lib/asterisk/sounds/en/. Is it possible to select different vm message based on certain conditions? I know I can play file with Playback() and then play empty vm-intro and I can also mess around with renaming files using System() however I was hoping there is a straight forward way rather than work-around. Thank you... Harel -- _ -- 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] Callback when available
Hi All, I would like to implement a call-back option when called user is busy. Consider this scenario: 1. A caller is calling a number which is busy on another call. 2. The system will prompt the caller (press 3 to be called back etc.) to be called back when called user is available. 3. Caller hangs up. problem: how to monitor called user status after calling user has hanged-up? Dialling plan has terminated at this point... 4. The called user terminates his previous call. 5. The system calls the caller and prompts him to wait for connection. 6. The system calls the called user and bridges the call upon pick up. I can use any version of Asterisk as required. Any opinions and ideas would be appreciated. Kind Regards, Harel This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee you should not disseminate or distribute a copy of this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi supp...@easycall.gi -- _ -- 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] 2nd network interface for RTP/media
Hi All, I would like to separate the media traffic from the signalling. Can Asterisk send and receive media (rtp) traffic from a secondary network interface? Thanks, Harel This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee you should not disseminate or distribute a copy of this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi supp...@easycall.gi -- _ -- 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] MoH and stuch channels
Hi All, I would like to separate the media traffic from the signalling. Can Asterisk send and receive media (rtp) traffic from a secondary network interface? Thanks, Harel This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee you should not disseminate or distribute a copy of this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi supp...@easycall.gi -- _ -- 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] MoH and stuch channels
Please ignore this message (wrong subject by mistake). Please see message with subject 2nd network interface for RTP/media Thanks Harel -- Message: 2 Date: Mon, 1 Nov 2010 12:52:16 +0100 From: Harel Cohen ha...@easycall.gi Subject: [asterisk-users] MoH and stuch channels To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Message-ID: cabc006234837141a279831f680414066b28f8d...@ecs.easycall.local Content-Type: text/plain; charset=us-ascii Hi All, I would like to separate the media traffic from the signalling. Can Asterisk send and receive media (rtp) traffic from a secondary network interface? Thanks, Harel * This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee you should not disseminate or distribute a copy of this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall's Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi supp...@easycall.gi -- _ -- 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] convert g729A-g729B and vice-versa
Hi all. Is there a free, or at least non-expensive, solution that can convert g729A --g729B (with VAD)? The no-support for g729B on Asterisk gives me a BIG headache… Thanks, Harel This electronic message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee you should not disseminate or distribute a copy of this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. EasyCall Ltd 11 Cornwall’s Parade P.O.Box 1488 Gibraltar. Office : +350 20077889 Fax : +350 20076727. www.easycall.gi supp...@easycall.gi -- _ -- 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] mapping of disconnect reasons
Sorry for the late response. Philipp, I've checked the file below and also the suggested voip-info link. None of those describe how or why Asterisk assumed that 402 should be mapped to NORMAL TERMINATION status. Both places refer to how Asterisk status should be mapped to SIP cause and not vice-versa. Could you (or someone) please take another look to locate the correct file? Thanks Harel -- Message: 4 Date: Wed, 04 Aug 2010 15:20:05 +0200 From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Subject: Re: [asterisk-users] mapping of disconnect reasons To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4c598525.14932.1d6...@klitzing.pool.informatik.rwth-aachen.de Content-Type: text/plain; charset=US-ASCII The mapping in Asterisk 1.4.24 is the problem: 402 Payment Required is mapped to 16 Normal termination instead of 21 Call Rejected. Could you direct me to the relevant file of code where these mappings are done? Before reporting a bug I would like to confirm whether this issue has been addressed on newer releases. Look in channels/chan_sip.c and search for 3398 See also: http://www.voip- info.org/wiki/index.php?page=Asterisk+variable+hangupcause Philipp -- -- _ -- 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] mapping of disconnect reasons
Tilghman, thank you for your reply. The mapping in RFC 3398 is logically correct therefore I do not need to submit a suggestion to its editor. The mapping in Asterisk 1.4.24 is the problem: 402 Payment Required is mapped to 16 Normal termination instead of 21 Call Rejected. Could you direct me to the relevant file of code where these mappings are done? Before reporting a bug I would like to confirm whether this issue has been addressed on newer releases. Thanks, Harel -- On Tuesday 03 August 2010 06:21:23 Philipp von Klitzing wrote: Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 Payment Required from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. * if you think the mapping is wrong, then you should open a ticket on the Asterisk bug tracker Actually, much of the mapping is specified by RFC 3398 section 8.2.6.1. Thus, if you think the mapping is wrong, you should submit a suggestion for amendment to the RFC editor. Only for response codes specified differently than in this section should you open an issue in the tracker. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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
[asterisk-users] mapping of disconnect reasons
Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 “Payment Required” from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is resulting call status “CONGESTION”) but will not do so for all normal terminations (16, Normal Termnation, 17 Busy, 18 No Answer). Thanks, Harel -- _ -- 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] perform tasks outside a dial-plan (not during a call)
Hi all, Can the Asterisk do “things” not during a call? For example I would like to change my dial plan during certain hours\dates or I would like to check some information in the astdb (e.g. counters of al sort) and handle it as required and so on. All of this is not call-related therefore I don’t know if I can somehow do it using the dial-plan applications\functions. I know I can do chron jobs on the Linux level but for maintenance and readability I would prefer to do these tasks from within the Asterisk. Is it possible to configure the Asterisk to perform routine tasks on certain times or certain intervals? Thanks, Harel -- _ -- 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] Incoming call doesn't finish when internal phone hangs up
Hi Elder, I would first check the behaviour of your PSTN lines (i.e. nothing to do with Asterisk). In many places PSTN companies allow between 30 to 90 seconds of connection to remain open even if the -called- party, NOT the calling party, has hung-up. Normally this is to allow putting down the phone in one room and picking up in another room without disconnecting the line. Make a small test to verify this and if this is the case you will need to discuss this with your PSTN provider. Harel Date: Thu, 8 Jul 2010 12:01:40 -0500 From: Daniel - Asterisk earohua...@gmail.com Subject: [asterisk-users] Incoming call doesn't finish when internal phone hangs up To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlktikxafnxhbsws0ov4u5ht3yjbeevuh26vehrg...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hello guys, I have this problem when a call is received in my PBX: (Caller) -- (Redirecting Service) -- (E1 PRI) -- (Asterisk PBX) -- (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me acording Caller's duration which is bigger than Asterisk's CDR. The same issue happens when Caller dials E1 PRI directly. In every case Asterisk finishes normally the call as CDR and CLI register correctly. I'm using Asterisk 1.4.21.2 and OpenVox DE210P card. Configuration files follow: zaptel.conf: span=1,1,1,ccs,hdb3 bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 span=2,2,1,ccs,hdb3 bchan=32-46,48-62 dchan=47 # Global data loadzone= us defaultzone = us zapata.conf: [channels] language=es context=default rxwink=300 usecallerid=yes hidecallerid=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 busydetect=yes busycount=yes busypattern=500,500 answeronpolarityswitch=yes hanguponpolarityswitch=yes ;PRI RDSI - SPAN 1 group = 1 context = incoming-1 inmediate=no switchtype=euroisdn signalling=pri_cpe channel = 1-15,17-31 ;PRI RDSI - SPAN 2 group = 1 context = incoming-2 inmediate=no switchtype=euroisdn signalling=pri_cpe channel = 32-46,48-62 ... Thanks in advance, Elder Arohuanca Lagos Phone: +51 1 991696900 Lima - Peru -- _ -- 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] Local channel usage
Hi All,
I’m trying to do “things” after my Dial application terminates (e.g. play IVR
to called party, calling party, etc.). I’m trying to use the local channel for
this purpose but so far with no success. I’m using 1.6.1.18 and this is my
extensions.conf:
[Internal]
exten = _22,1,Dial(Local/${ext...@cw/n) ; 22 is test number
exten = _22,2,Noop(After Hangup)
[CW]
exten = _x.,1,Dial(SIP/307)
exten = _x.,2,Noop(After Hangup)
The call never reaches neither of the Noop applications. Consol:
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [...@internal:1] Dial(SIP/309-00a5, Local/2...@cw/n)
in new stack
-- Called 2...@cw/n
-- Executing [...@cw:1] Dial(Local/2...@cw-af6f;2, SIP/307) in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called 307
-- SIP/307-00a6 is ringing
-- Local/2...@cw-af6f;1 is ringing
-- SIP/307-00a6 is ringing
-- SIP/307-00a6 is ringing
-- SIP/307-00a6 is ringing
-- SIP/307-00a6 answered Local/2...@cw-af6f;2
-- Local/2...@cw-af6f;1 answered SIP/309-00a5
== Spawn extension (CW, 22, 1) exited non-zero on 'Local/2...@cw-af6f;2'
== Spawn extension (Internal, 22, 1) exited non-zero on 'SIP/309-00a5'
If I use the ‘g’ option in my Dial() both Noop will be run only if the called
party hang-up first. I need a simple continuation in the dial plan regardless
of who disconnected the call.
Thanks in advance
Harel
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Re: [asterisk-users] Local channel usage
Zeeshan: 1. g option continues the dial plan after the called party hangup, and only the called party. See the manual or check for yourself... 2. h extension is no good for me because the voice path is already closed at this point therefore I cannot play IVR (Im getting Warnings like: file.c:750 ast_readaudio_callback: Failed to write frame). Tiago: There is no Dial() option to simply continue dial-plan after Dial(). See above regarding g option. Can anyone think of a way to play IVR after conversation initiated by Dial() terminates? Harel -- Message: 9 Date: Tue, 22 Jun 2010 07:27:42 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Local channel usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlktilo2hzyq4jp7rb_iguzaq-n2chxnhq96grg0...@mail.gmail.com Content-Type: text/plain; charset=windows-1252 'g' option continues the dial plan after the call has been answered, not after it is hung up. Depending upon what you are trying to do, first try to use h extension, i.e. in the example you gave, you should replace '_22,2' with 'h,1'. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-22 6:23 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hi, After a Dial, the call is hung up. It doesn't carry on with dialplan unless you specify the appropriate dial option. Check wiki voip-info for cmd Dial, I think the option is g 2010/6/22 Harel Cohen ha...@easycall.gi Hi All, I?m trying to do ?things? after my Dial application terminates (e.g. play IVR to cal... -- _ -- 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] conf files vs astdb
Hi all,
Could someone please tell me what is the relative cost in using conf files
oppose to the astdb? Basically I need to match a name to a phone number in
order to have all users registered by name and not by number (which I
understood is not a good practice). I have 2000 users and a complex dial-plan
and server resources become an issue.
I could implement this via a context in my extensions.conf:
exten = number,1,Dial(SIP/name) ; obviously I would need to hard-code this
for every extension
- or I could do it via astdb:
exten = _XXX,1,Dial(SIP/${DB(Names/${EXTEN})})
Which method would consume fewer resources (put aside other pro's con's)?
Is there any better way of implementing this?
Would 'hints' help me out here? If yes, I would appreciate a detailed
explanation how to use it.
Thanks in advance,
Harel
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[asterisk-users] Call-Waiting, implementation ideas
Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk) SomeuserY is calling MyUserA. SomeuserY gets a special ringing tone. Meaning - Asterisk opens voice channel towards SomeuserY (progress with SDP) and plays SpecialRingBack.wav/gsm etc. MyUserA Gets voice notification (e.g. beep-beep) during his call to SomeuserX. Meaning - Asterisk barge-in the rtp stream and play the file beepbeep.wav/gsm on the MyUserA channel. This is done periodically for as long as SomeuserY is waiting to be answered (i.e. doesn't hang-up). Asterisk is monitoring the state of the call SomeuserX - MyUserA. If MyUserA will signal (e.g. hook-flash or some digit sequence) that he wants to answer the 2nd call then Asterisk will put on hold SomeuserX and bridge SomeuserY to MyUserA with the option for MyUserA to toggle between the two channels. If the conversation SomeuserX with MyUserA is terminated Asterisk will INVITE MyUserA and when picked up will bridge SomeuserY with MyUserA. I hope there is a solution for that… I tried using DEVICE_STATE for this purpose however I keep getting status NOT_INUSE even if the extension IS in use (I'll open a different thread on this issue if needed). Thanks in advance for any ideas provided, Harel -- _ -- 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] G729 exhaustion conditions
Hi all, Suppose I buy and install one G729 codec. Suppose there is one call going on where both end-points have G729 codecs and the Asterisk is not doing any transcoding. Does this conversation exhaust my G729 license (even though this call would have worked without license in the first place) or do I still have the ability to use this G729 codec for other call which requires transcoding? Thank you, Harel Cohen -- _ -- 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
