[asterisk-users] Local channel that runs a custom app... why immediate hangup?
I have an app which handles a Mitel's command port to change the MWI
lights. It detects dial tone, plays some DTMF digits, listens for
dialtone-or-busy, does a manager event on what it finds, and returns.
Since the Mitel command port does not give answer supervision (looks like
it's ringing), and since I run this app via a AMI originate command, I set
up an extension in extensions.conf thus:
exten = MWISend,1,Answer(1000) ; does more-or-less the same thing without
the delay
exten = MWISend,2,MWISend(${MWISEND_DIGITS}) ; the variable set in the
AMI call - this is tested works.
The Originate goes like:
Channel: Local/mwis...@default
Extension: 650nnn
Context: direct_out ; a context that dials SIP/${ext...@metaswitch which
is known to work
Priority: 1
What happens is the local channel goes immediately into Hangup with many
odd-looking messages about searching for the extension, when it has clearly
already been found and goes into a hangup(see trace below).
Anybody have a hint what's happening here and how to make the application
see the voice frames from the dial-out?
Thanks.
[Oct 30 23:44:45] DEBUG[16246] manager.c: Manager received command
'originate'
[Oct 30 23:44:45] DEBUG[16247] pbx.c: Launching 'Answer'
[Oct 30 23:44:45] DEBUG[16247] devicestate.c: Notification of state change
to be queued on device/channel Local/mwis...@default
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: No provider found, checking
channel drivers for Local - mwis...@default
[Oct 30 23:44:45] DEBUG[16246] devicestate.c: Notification of state change
to be queued on device/channel Local/mwis...@default
[Oct 30 23:44:45] DEBUG[16150] chan_local.c: Checking if extension
mwis...@default exists (devicestate)
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: Changing state for
Local/mwis...@default - state 2 (In use)
[Oct 30 23:44:45] DEBUG[16176] app_queue.c: Device 'Local/mwis...@default'
changed to state '2' (In use) but we don't care because they're not a member
of any queue.
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: No provider found, checking
channel drivers for Local - mwis...@default
[Oct 30 23:44:45] DEBUG[16150] chan_local.c: Checking if extension
mwis...@default exists (devicestate)
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: Changing state for
Local/mwis...@default - state 2 (In use)
[Oct 30 23:44:45] DEBUG[16176] app_queue.c: Device 'Local/mwis...@default'
changed to state '2' (In use) but we don't care because they're not a member
of any queue.
[Oct 30 23:44:45] DEBUG[16248] channel.c: Soft-Hanging up channel
'Local/mwis...@default-1b18,1'
[Oct 30 23:44:45] DEBUG[16248] channel.c: Hanging up channel
'Local/mwis...@default-1b18,1'
[Oct 30 23:44:45] DEBUG[16247] pbx.c: Spawn extension (default,MWISend,1)
exited non-zero on 'Local/mwis...@default-1b18,2'
[Oct 30 23:44:45] DEBUG[16247] channel.c: Soft-Hanging up channel
'Local/mwis...@default-1b18,2'
[Oct 30 23:44:45] DEBUG[16247] channel.c: Hanging up channel
'Local/mwis...@default-1b18,2'
[Oct 30 23:44:45] DEBUG[16247] devicestate.c: Notification of state change
to be queued on device/channel Local/mwis...@default
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: No provider found, checking
channel drivers for Local - mwis...@default
[Oct 30 23:44:45] DEBUG[16150] chan_local.c: Checking if extension
mwis...@default exists (devicestate)
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: Changing state for
Local/mwis...@default - state 2 (In use)
[Oct 30 23:44:45] DEBUG[16176] app_queue.c: Device 'Local/mwis...@default'
changed to state '2' (In use) but we don't care because they're not a member
of any queue.
[Oct 30 23:44:45] DEBUG[16248] devicestate.c: Notification of state change
to be queued on device/channel Local/mwis...@default
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: No provider found, checking
channel drivers for Local - mwis...@default
[Oct 30 23:44:45] DEBUG[16150] chan_local.c: Checking if extension
mwis...@default exists (devicestate)
[Oct 30 23:44:45] DEBUG[16150] devicestate.c: Changing state for
Local/mwis...@default - state 1 (Not in use)
[Oct 30 23:44:45] DEBUG[16176] app_queue.c: Device 'Local/mwis...@default'
changed to state '1' (Not in use) but we don't care because they're not a
member of any queue.
--- at this point the AMI application sees the hangup event and terminates
---
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Re: [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working
Joseph wrote: I always had a problem with SIP and DTMF, I'm using old sipura adapters and have one digium iaxy FXS unit which works almost perfectly, never had any problem with DTMF on this unit. However, all phones connected to Sipura don't work very well especially when I setup speed dialing numbers to do banking, entering account number etc. (sipura units are set to dtmf=auto) DTMF Detection is done at your ATA, asterisk has nothing to do with that. Try the different settings on your sipura unit to improve DTMF detection, ie: relaxed dtmf, rx/tx gain, impedance. I don't remember exaclty but you don't need auto if you are certain that your SIP endpoint is either info or rfc2833. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI/ZAP overlap dialing
On Fri, Oct 30, 2009 at 04:54:46AM -0700, Vieri wrote:
Hi,
I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
I'm having some trouble with overlap dialing.
Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an
Alcatel prefix of type ARS Prof.Trg Grp Seiz.with overlap.
I'm expecting Asterisk to receive '1004053' (where '100' is a prefix which
always shows up in the euroisdn setup).
However, Asterisk is only receiving '1004' which means that it's not reading
the digits that follow.
Are there issues with receiving overlap dials from zap channels?
According to the Alcatel trace below, it looks like Asterisk is accepting the
call before a Sending complete is released by Alcatel.
I'm using libpri 1.2.8 and Asterisk 1.2.31.1.
I'm not sure if handling of overlap hasn't changed since.
But can you provide a trace of how Asterisk sees things? e.g. 'pri
intense debug span 1'
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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[asterisk-users] OT - Number Portability
Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural monopoly. From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them, and a set of DID numbers we purchase from them. Do we have to be a full CLEC to participate as a carrier? Does this imply we must have an SS7 connection to the PSTN? Thanks for any info, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI/ZAP overlap dialing
On Sat, Oct 31, 2009 at 5:27 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: I'm not sure if handling of overlap hasn't changed since. But can you provide a trace of how Asterisk sees things? e.g. 'pri intense debug span 1' the intense debug is overkill we only need messages of layer 3 ... just do pri debug span 1 Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Number Portability
Your chances are likely slim to none. But good luck. First to port numbers you have to be a recognized carrier, which for the most part means getting numbers from NANPA : North American Numbering Plan Administration. To do that you have to be certified by your state PUC or be a CMRS (cell phone) carrier. They would give you a block of 10,000 numbers designated to the rate center of the ILEC in question. Then you designate on of those numbers as a local routing number (LRN) which is like a pathfinder number for ported numbers. And, you work out an Interconnection agreement with the local Telco (probably with them kicking and screaming for months or a year) because they really don't want you there, and you aren't a big cell phone company, but a local wire line competitor, which then is approved by the state PUC. What some others have done is to operate as a PBX Service provider or some other business term. They get a PRI from the local company, and become the agent for the customer, move the service delivery to their PRI, and then distribute the calls to the appropriate customer via SIP and Asterisk or other solution. That has worked in Casa Grande, AZ for one place. (Not ours.) Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Saturday, October 31, 2009 8:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT - Number Portability Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural monopoly. From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them, and a set of DID numbers we purchase from them. Do we have to be a full CLEC to participate as a carrier? Does this imply we must have an SS7 connection to the PSTN? Thanks for any info, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Number Portability
Two more comments. Yes, to join the PSTN call distribution system you must have SS7. While rural ILECs are not exempt from number portability, there is a court injunction that saves them from having to transport the call out of their local rate center, so getting calls from a distant RILEC to a central point is at a cost to the requesting carrier. There are other complexities. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Saturday, October 31, 2009 8:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT - Number Portability Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural monopoly. From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them, and a set of DID numbers we purchase from them. Do we have to be a full CLEC to participate as a carrier? Does this imply we must have an SS7 connection to the PSTN? Thanks for any info, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic
Thanks, I tried these options with no luck. I have an RMA in place for the card. Tried loading a fresh install of Centos with no change. Will try another card and hopefully try this card in another machine. On Oct 30, 2009, at 5:49 PM, Mariano Lecuona wrote: Take a look at this document. This may help you on trouble shoot your kernel panic. http://www.novavox.co.uk/docs/install-guides/novavox-asterisk-card-installation-issues.pdf 2009/10/30 David Shauger sollost...@gmail.com Setting up a new Asterisk server with Centos 5.2 and Asterisk 1.4.23 using Dahdi and getting a kernel panic - not syncing: Fatal exception error during boot. Anyone have thoughts on what I can do to rectify this or is this card not compatible with this machine? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Number Portability
On Sat, 31 Oct 2009, Cary Fitch wrote: Two more comments. Yes, to join the PSTN call distribution system you must have SS7. While rural ILECs are not exempt from number portability, there is a court injunction that saves them from having to transport the call out of their local rate center, so getting calls from a distant RILEC to a central point is at a cost to the requesting carrier. There are other complexities. Cary Fitch Thanks Cary! We figured as much, and I appreciate the confirmation. We are not afraid of the battle and will be heading into this anyway. Feel a bit like Don Quixote :) j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disconnecting during the call, analog lines
Hi All; Asterisk version is 1.6.1.8 Dahdi version is: dahdi-linux-complete-2.2.0.2+2.2.0 Since long time, and I am facing this problem and I did all the trouble shooting that I know without any success. The problem that while we are talking with someone through the FXO (connected to the PSTN analoge line), suddenly the call disconnect (without any specific time). I tried callprogress=no and still the problem happens. What are the solutions to resolve this problem? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls disconnects after short time
Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP/213.165.32.100-b7d21018' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7d21018, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7d21018, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7d21018, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7d21018, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7d21018, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7d21018, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7d21018, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7d21018' elastix*CLI ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] !command from Manager
Hi, Is it possible to run a !command from Manager connection? Thanks in advance! CB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-1.6.1.8 DTMF with SIP is not working
On 10/31/09 10:24, Ivan Stepaniuk wrote: Joseph wrote: I always had a problem with SIP and DTMF, I'm using old sipura adapters and have one digium iaxy FXS unit which works almost perfectly, never had any problem with DTMF on this unit. However, all phones connected to Sipura don't work very well especially when I setup speed dialing numbers to do banking, entering account number etc. (sipura units are set to dtmf=auto) DTMF Detection is done at your ATA, asterisk has nothing to do with that. Try the different settings on your sipura unit to improve DTMF detection, ie: relaxed dtmf, rx/tx gain, impedance. I don't remember exaclty but you don't need auto if you are certain that your SIP endpoint is either info or rfc2833. That could be part of the problem as well. The old sipura 3K (green box) doesn't have setting: relaxed dtmf (the newer silver one Linksys do but the echo on PSTN line is unacceptable, and almost impossible to fix, but this is another problem). On the sipura 3k ATA I've try to change setting from DTMF=Auto to DTMF-AVT (that is rfc2833) but it didn't help. I have two standard analog phones both Uniden (but different models) one is transmitting DTMF OK via both Sipur ATA and Digium iaxy + Sipura ATA (PSTN) swapping the other phone fails on both lines. So it could be the problem with the phone but I have tried two other different ones and both fail to transmit DTMF correctly. Adjusting the gain on PSTN line (+3 -3) doesn't make any difference. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI line resetting on incoming call
Was'nt sure if this mail got through earlier: I have been having a weird issue with my telco's ISDN PRI occasionally resetting on a incoming call, i suspect it to possibly be a timing issue since some of the incoming call work. This problem happens very frequently. I am using asterisk-1.6.0.1 with libpri-1.4.9, the asterisk server is connected viw TDMoE to a Redfone Fonebridge into which my telco's ISDN PRI line is connected. I have used the exact same setup in another office and it worked seamlessly. After setting it up, i would notice every couple of minutes the entire line would reset usually timed with a incoming call, i tried getting help from my telco, but they were completely incompetent. Before deploying the asterisk solution the PRI was hooked up to an hardware PBX, and the line worked fine, even now when i hook it up to the hardware pbx is works great, but when connected to the asterisk server we start to see these disconnects. Following are some of the pertinent sections of my chan_dadhi dahdi/system.conf: dahdi/system.conf # spans dynamic=ethmf,eth1/00:50:C2:65:D6:54/0,31,2 dynamic=ethmf,eth1/00:50:C2:65:D6:54/1,31,1 # Channels bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 alaw=1-62 # Tonezone loadzone=in defaultzone=in chan_dahdi.conf [channels] pridialplan=unknown overlapdial=yes usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no group=1 callgroup=1 pickupgroup=1 group=2 context=autoatt switchtype=national signalling=pri_cpe musiconhold=default channel=1-15,17-31 useincomingcalleridondahditransfer = yes group=3 context=hardpbx switchtype=euroisdn signalling=pri_net musiconhold=default channel=32-46,48-62 useincomingcalleridondahditransfer = yes I began to look at the ISDN debug output to try and determine what was causing the line to reset. After a few days of pouring over the output i noticed a pattern: [ 02 01 14 16 08 02 06 1e 4d ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 010 0: 0 N(R): 011 P: 0 5 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 10 to (but not including) 11 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 1566/0x61E) (Originator) Message type: RELEASE (77) -- Making new call for cr 1566 [ 00 01 16 16 08 02 86 1e 5a 08 02 81 d1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 011 0: 0 N(R): 011 P: 0 9 bytes of data Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1566/0x61E) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (e.g. parameter out of range) (5) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Restarting T203 timer [ 00 01 01 18 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 012 P/F: 0 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 10 to (but not including) 12 -- ACKing packet 11, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Restarting T203 timer q921.c:842 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED I would see the RELEASE message, and then a 'Making new call' indication following which i would see a RELEASE COMPLETE message with ' Invalid call reference value' and then the line resets. Any idea why this would happen..? Thanks sam!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?
Hi actually, i test a new Asterisk Server and i want add Mysql Realtime SIP. I read on the wiki: === Database Config put the following in res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = myuser dbpass = mypass dbport = 3306 Values in sip.conf or iax.conf like in older versions of * are no longer used. Database Table Lets create the table we need: NOTE: You can use any table name you wish, just make sure the table name matches what you have the family name bound to. === But i don't see where i put the Table Name ? (if i don't want use sip_buddies) and he have a sample of Table Structure, can i add a new champs for my personnal software without problems ? Thanks jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls disconnects after short time
Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP/213.165.32.100-b7d21018' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7d21018, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7d21018, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7d21018, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7d21018, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7d21018, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7d21018, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7d21018, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7d21018' elastix*CLI ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Long pause during dialing to IVR
To insert long pause during dialing and submitting multiple DTMF tones, is there better solution then below: exten = _51,1,Dial(SIP/18778794...@pstn-5665,300,D(www1www),D(005893884053811#)) I think submitting multiple DTMF tones is not allow from one command line. The first part D(www1www) worked, but not the second one D(005893884053811#) I'm trying to obtain credit card authorization number from Bank's IVR So dialing banks phone number, Press: 1 enter merchant: device #1 enter merchant: device #2 enter 1 enter 2 enter credit_card_number (most likely from external input file) enter enter expiry_date (from external input file) enter amount: manually. I'm sure somebody has written something like this before. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [IAX] Recommended soft- and hardphones?
On 10/30/09 12:55, Vincent wrote: Hello Since SIP/RTP is a pain to use with road warriors who need to connect from any location over the Internet, I'd like to get them some IAX phones instead. For those of you using this protocol instead of SIP, what would you recommend as IAX hardphones and Windows (and ideally Mac) softphones? How about Digium iaxy adapter, I've used it in the past, it register to your asterisk as soon as you plug it to any network (borrow any hotels phone, plug it into the iaxy adapter) and you have your solution. The is a web-page that will allow you to provision the adapter over the Internet if you have to (don't have the link). -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic
For anyone interested, this is an HP ML115 Proliant server using AMD. We put in a PCI Digium card and all was bliss. Also found the PCI Express card works fine in a Dell T100 with Xeon processors. On Oct 30, 2009, at 5:49 PM, Mariano Lecuona wrote: Take a look at this document. This may help you on trouble shoot your kernel panic. http://www.novavox.co.uk/docs/install-guides/novavox-asterisk-card-installation-issues.pdf 2009/10/30 David Shauger sollost...@gmail.com Setting up a new Asterisk server with Centos 5.2 and Asterisk 1.4.23 using Dahdi and getting a kernel panic - not syncing: Fatal exception error during boot. Anyone have thoughts on what I can do to rectify this or is this card not compatible with this machine? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnecting during the call, analog lines
On 10/31/09 08:20, bilal ghayyad wrote: Hi All; Asterisk version is 1.6.1.8 Dahdi version is: dahdi-linux-complete-2.2.0.2+2.2.0 Since long time, and I am facing this problem and I did all the trouble shooting that I know without any success. The problem that while we are talking with someone through the FXO (connected to the PSTN analoge line), suddenly the call disconnect (without any specific time). I tried callprogress=no and still the problem happens. What are the solutions to resolve this problem? Have you tried running debug mode on the CLI, I'm sure it will print something after disconnecting; was it happening in previous version as well, did you change anything in a dial plan etc? -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls disconnects after short time
My server use public ip, so no nat issues, here is the out of sip debug: - --- (10 headers 0 lines) --- Sending to 213.165.32.100 : 5060 (no NAT) --- Reliably Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 elastix*CLI --- Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- Hungup 'IAX2/9-4490' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966599740196, 4) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7c10ad8, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7c10ad8, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7c10ad8, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7c10ad8, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' elastix*CLI thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP/213.165.32.100-b7d21018' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7d21018, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7d21018, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7d21018, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7d21018, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7d21018, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7d21018, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7d21018, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on
Re: [asterisk-users] Long pause during dialing to IVR
On 10/31/09 12:56, Joseph wrote: To insert long pause during dialing and submitting multiple DTMF tones, is there better solution then below: exten = _51,1,Dial(SIP/18778794...@pstn-5665,300,D(www1www),D(005893884053811#)) I think submitting multiple DTMF tones is not allow from one command line. The first part D(www1www) worked, but not the second one D(005893884053811#) I'm trying to obtain credit card authorization number from Bank's IVR So dialing banks phone number, Press: 1 enter merchant: device #1 enter merchant: device #2 enter 1 enter 2 enter credit_card_number (most likely from external input file) enter enter expiry_date (from external input file) enter amount: manually. It could be similar to this one: exten = _51,1,Dial(SIP/18778794...@pstn-5665,300,D(w1w)) exten = _51,n,Wait(4) exten = _51,n,SendDTMF(005893884053811#) exten = _51,n,Wait(4) exten = _51,n,SendDTMF(123456#) but only the first one go through. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID
On 10/30/09 10:32, Carlos Chavez wrote: On Fri, 2009-10-30 at 08:37 -0500, Tilghman Lesher wrote: On Thursday 29 October 2009 12:32:48 Carlos Chavez wrote: On Thu, 2009-10-29 at 12:23 -0500, Tilghman Lesher wrote: On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote: On Wed, 2009-10-28 at 23:36 -0500, Tilghman Lesher wrote: On Wednesday 28 October 2009 17:57:49 Carlos Chavez wrote: I am having a problem with Asterisk 1.6.2.0-rc3 and Asterisk-Addons 1.6.2.0-rc1 when recording CDR to a Mysql database. All fields except callerid are recorded properly after every call. I have both a clid and callerid field in the database but both fields are empty. In cdr_mysql.conf I have this alias in the [columns] section: alias start = calldate alias callerid = clid Get rid of this alias callerid = clid line. What it does is to tell the driver to put the CDR variable called callerid into the clid column in the database, overriding the builtin clid mapping. Then reload. If you want the Caller*ID information in the callerid column, then your mapping is backwards and should be alias clid = callerid. Remember, the arrow points in the direction that the information flows: FROM the cdr TO the database. I already tried that with the same result. I even added a callerid column to my cdr table just in case. Either removing the alias line or reversing it like you suggested will not record the callerid in either column. Try the following commands. What is output? CLI core set debug 1 CLI module reload cdr_addon_mysql.so Just this: pbxoficina*CLI core set debug 1 Core debug is at least 1 pbxoficina*CLI module reload cdr_addon_mysql.so -- Reloading module 'cdr_addon_mysql.so' (MySQL CDR Backend) pbxoficina*CLI Do you have debug set to go to console in logger.conf? Yes: pbxoficina*CLI logger show channels Channel Type StatusConfiguration --- --- /var/log/asterisk/messages File Enabled- Warning Notice Error Console Enabled- Debug Warning Did you solved this problem? I'm having the same issue on asteriks-1.6.1.8 except in my case no records are being passed to mysql. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Determining extension's sip.conf default mailbox
Hello list,
How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with sip show peer ext)? Is there a
function to extract it?
Why? Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the extension's default
mailbox as coded in sip.conf.
sip.conf
--
[100]
mailbox=100
[102]
mailbox=102
[103]
mailbox=100
I want a function which I can use in the dialplan (1.6) that works like:
DefaultMailbox(100) - 100
DefaultMailbox(102) - 102
DefaultMailbox(103) - 100
for example:
exten s,n,VoicemailMain(DefaultMailbox(${CALLERID(num)}))
Suggestions?
Thanks!
___
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Re: [asterisk-users] !command from Manager
On Sat, Oct 31, 2009 at 12:04:18PM -0400, cbulist wrote: Hi, Is it possible to run a !command from Manager connection? No. You can implement it yourself. '!' is not sent to the asterisk daemon. Rather, the local client runs a command. For instance: # id -a uid=0(root) gid=0(root) groups=0(root) # ps u `cat /var/run/asterisk/asterisk.pid ` USER PID %CPU %MEMVSZ RSS TTY STAT START TIME COMMAND asterisk 4314 0.0 0.2 698424 5036 ?Ssl Oct10 17:53 /usr/sbin/aster # asterisk -r Asterisk 1.6.2.0~dfsg~beta4-0.7501, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = This package has been modified for the Debian GNU/Linux distribution Please report all bugs to http://bugs.debian.org/asterisk = Connected to Asterisk 1.6.2.0~dfsg~beta4-0.7501 currently running on sweetmorn (pid = 4314) sweetmorn*CLI !id -a uid=0(root) gid=0(root) groups=0(root) That said, the dialplan application System allows you to do that. E.g. look for the dialplan snippet that includes the extension called 'executecommand' which is embedded in http://svn.digium.com/svn/asterisk-gui/branches/2.0/config/js/pbx.js Needless to say that this opens the door to shell code injection attacks, such as the one described in http://www.csnews.com/csn/news/article_display.jsp?vnu_content_id=1004015447 Actually http://en.wikipedia.org/wiki/Code_injection#Shell_injection will probably be more useful. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls disconnects after short time
The only informative part are the 2 paragraphs of the sip debug, but can't tell much since you only show a very small portion of the sip log. There is a 487 Request terminated there screaming at you but can't tell if meaning that provider is not handling the ACKs. That section of the [macro-hangupcall] context is useless as it is caused by the hangup, and not an effect. The usage of a public IP is not indicative of the existence of a firewall which can be blocking any necessary ports for tcp and/or udp. You should always cover your real IP numbers when showing samples of your logs CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Sunday, November 01, 2009 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time My server use public ip, so no nat issues, here is the out of sip debug: - --- (10 headers 0 lines) --- Sending to 213.165.32.100 : 5060 (no NAT) --- Reliably Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 elastix*CLI --- Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- Hungup 'IAX2/9-4490' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966599740196, 4) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7c10ad8, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7c10ad8, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7c10ad8, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7c10ad8, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' elastix*CLI thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on
Re: [asterisk-users] Calls disconnects after short time
Hello, I have grabbed again a whole call when it hangs up debug, I dono what else I can read?? What exactly you want me to look for? And assuming there is a firewall at my ISP, how to diagnose it? Thanks for the advise, Here is another log: -- Called 9/0557202919 -- Call accepted by xxx.xxx.xxx.xxx (format ulaw) -- Format for call is ulaw elastix*CLI --- SIP read from xx.xx.xx.xx:5060 --- CANCEL sip:966557202...@xx.xx.xx.xx SIP/2.0 Max-Forwards: 70 To: 966557202919 sip:966557202...@xx.xx.xx.xx From: sip:9998...@xx.xx.xx.xx;tag=3466014864-147468 Call-ID: 19773310-3466014864-147...@aaa.bbb.net CSeq: 1 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec Contact: sip:9998...@xx.xx.xx.xx:5060 Content-Length: 0 - --- (10 headers 0 lines) --- Sending to xx.xx.xx.xx : 5060 (no NAT) --- Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060 --- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx. xx.xx.xx From: sip:9998...@xx.xx.xx.xx;tag=3466014864-147468 To: 966557202919 sip:966557202...@xx.xx.xx.xx;tag=as717c0994 Call-ID: 19773310-3466014864-147460@ aa.bb.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Transmitting (no NAT) to xx.xx.xx.xx:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx. xx.xx.xx From: sip:9998...@xx.xx.xx.xx;tag=3466014864-147468 To: 966557202919 sip:966557202...@xx.xx.xx.xx;tag=as717c0994 Call-ID: 19773310-3466014864-147...@aa.bb.net CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- Hungup 'IAX2/9-8610' Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 4:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time The only informative part are the 2 paragraphs of the sip debug, but can't tell much since you only show a very small portion of the sip log. There is a 487 Request terminated there screaming at you but can't tell if meaning that provider is not handling the ACKs. That section of the [macro-hangupcall] context is useless as it is caused by the hangup, and not an effect. The usage of a public IP is not indicative of the existence of a firewall which can be blocking any necessary ports for tcp and/or udp. You should always cover your real IP numbers when showing samples of your logs CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Sunday, November 01, 2009 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time My server use public ip, so no nat issues, here is the out of sip debug: thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determining SIP peer's default mailbox
Steve Johnson schrieb:
How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with sip show peer ext)? Is there a
function to extract it?
Why? Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the extension's default
mailbox as coded in sip.conf.
sip.conf
--
[100]
mailbox=100
[102]
mailbox=102
[103]
mailbox=100
I want a function which I can use in the dialplan (1.6) that works like:
DefaultMailbox(100) - 100
DefaultMailbox(102) - 102
DefaultMailbox(103) - 100
for example:
exten s,n,VoicemailMain(DefaultMailbox(${CALLERID(num)}))
SIPPEER(...|mailbox) I guess.[1] E.g.
VoicemailMain(${SIPPEER(${CALLERID(num)}|mailbox)});
[1] http://www.das-asterisk-buch.de/2.1/functions-sippeer.html
Philipp Kempgen
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[asterisk-users] need help debug asterisk-1.6 sip connection
I have a DID but for some reason is not working in asterisk-1.6 The same sip connection in asterisk-1.4 is working OK, but it doesn't work with asterisk-1.6 Here is my sip.conf section: ... [actio-out] type=friend secret=password user=48746612254 username=48746612254 fromuser=48746612254 authname=48746612254 callerpage=48746612254 fromdomain=sip.actio.pl host=sip.actio.pl insecure=very nat=yes qualify=yes dtmfmode=inband disallow=all allow=ulaw allow=alaw context=internal canreinvite=no Here is relevant section from asterisk-1.6 (failed connection) and asterisk-1.4 (working connection) == start asterisk-1.6 (not working) == - --- (17 headers 18 lines) --- == Using SIP RTP CoS mark 5 Sending to 81.15.150.20 : 5060 (no NAT) Using INVITE request as basis request - ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o Found peer 'actio-out' for '17804791270' from 81.15.150.20:5060 --- Reliably Transmitting (NAT) to 81.15.150.20:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK4c28.d397a70c58c5c983c7d85bb171d8e3b2.0;received=81.15.150.20 Via: SIP/2.0/UDP 81.15.150.20:5061;branch=z9hG4bKba07785b184a5f79266bde33dccc8212;rport=5061 From: sip:17804791...@81.15.150.20;tag=26a9eb26114a01c9f4d1f64b72cc1d9e To: sip:48746612...@81.15.150.20;tag=as52ab0bbb Call-ID: ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o CSeq: 200 INVITE Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0da18b05 Content-Length: 0 Scheduling destruction of SIP dialog 'ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o' in 13632 ms (Method: INVITE) syscon2*CLI --- SIP read from UDP://81.15.150.20:5060 --- ACK sip:s...@68.148.245.78:61454 SIP/2.0 Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK4c28.d397a70c58c5c983c7d85bb171d8e3b2.0 From: sip:17804791...@81.15.150.20;tag=26a9eb26114a01c9f4d1f64b72cc1d9e Call-ID: ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o To: sip:48746612...@81.15.150.20;tag=as52ab0bbb CSeq: 200 ACK User-Agent: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 - --- (8 headers 0 lines) --- syscon2*CLI --- SIP read from UDP://81.15.150.20:5060 --- = end asterisk-1.6 (not working) = == start asterisk-1.4 (working) == - --- (17 headers 18 lines) --- Sending to 81.15.150.20 : 5060 (no NAT) Using INVITE request as basis request - f203cdef-c5d411de-932ae4a5-81fb2...@82.177.2.12~1o Found peer 'actio-out' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 4 Found RTP audio format 98 Found RTP audio format 99 Found RTP audio format 2 Found RTP audio format 100 Peer audio RTP is at port 81.15.150.20:46648 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G723 for ID 4 Found unknown media description format G726-16 for ID 98 Found unknown media description format G726-24 for ID 99 Found audio description format G726-32 for ID 2 Found unknown media description format X-NSE for ID 100 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 81.15.150.20:46648 Looking for s in from_poland (domain 68.148.245.78) list_route: hop: sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr syscon4*CLI --- Transmitting (NAT) to 81.15.150.20:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK5978.8397fd91b29a224fb6158a2eb64d4489.0;received=81.15.150.20 Via: SIP/2.0/UDP 81.15.150.20:5061;branch=z9hG4bKa185fc54438defa99101bdc43db8e8c7;rport=5061 Record-Route: sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr From: sip:17804791...@81.15.150.20;tag=1f5a641fc6ffb42064d4123781f0e7bb To: sip:48746612...@81.15.150.20 Call-ID: f203cdef-c5d411de-932ae4a5-81fb2...@82.177.2.12~1o CSeq: 200 INVITE User-Agent: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:s...@10.0.0.109 Content-Length: 0 -- Executing [...@from_poland:1] Answer(SIP/48746612254-00789120, ) in new stack Audio is at 10.0.0.109 port 13414 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP --- Reliably Transmitting (NAT) to 81.15.150.20:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK5978.8397fd91b29a224fb6158a2eb64d4489.0;received=81.15.150.20 Via: SIP/2.0/UDP 81.15.150.20:5061;branch=z9hG4bKa185fc54438defa99101bdc43db8e8c7;rport=5061 Record-Route: sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr
