[asterisk-users] Peers Listed in sip show channels
Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : sip show channels [trixbox ~]# /usr/sbin/asterisk -rx sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/0 0x0 (nothing)No 192.168.1.116(None) YTc4ZmM3NjV 00101/6 0x0 (nothing) No Rx: REGISTER 195.189.173.10 301241893b37329b407 18996/0 0x0 (nothing)No 192.168.1.13 10072da66c6d6a1 00102/0 0x280100 (g729| No Tx: ACK 192.168.1.13 100567384261131 00102/0 0x280100 (g729| No Tx: ACK 192.168.1.13 1010041c9a77455 00102/0 0x280100 (g729| No Tx: ACK 81.201.84.45 3473290576 PUM273-UMU5 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 2706513184 ISB67X-ZJQN 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 4023308836 G7JP5O-AA4J 00101/00102 0x100 (g729) No Rx: ACK 192.168.1.13 10160758ea9a349 00102/0 0x0 (nothing)No (d) Tx: ACK 122.169.113.145 1006379b29497d0 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 10063a4fc558695 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 10067678828b011 00102/0 0x0 (nothing) No Init: NOTIFY 13 active SIP channels The last 3 rows have been there since past 6 days. There is no user 1006, logged into the system... I have 2 questions: (1) Where does Trixbox store this information (2) How can I periodically remove these records Thx in advance. Sanjay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peers Listed in sip show channels
do you have that user 1006 defined by IP ? does it have mailbox= also defined ? my wild guess is that there's unchecked voicemail and asterisk tries to initialize sending NOTIFY MWI messages you can't remove these messages they remove themselves after some timeout Martin On Sun, Sep 27, 2009 at 1:00 AM, RSCL Mumbai rscl.mum...@gmail.com wrote: Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : sip show channels [trixbox ~]# /usr/sbin/asterisk -rx sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/0 0x0 (nothing) No 192.168.1.116 (None) YTc4ZmM3NjV 00101/6 0x0 (nothing) No Rx: REGISTER 195.189.173.10 30124189 3b37329b407 18996/0 0x0 (nothing) No 192.168.1.13 1007 2da66c6d6a1 00102/0 0x280100 (g729| No Tx: ACK 192.168.1.13 1005 67384261131 00102/0 0x280100 (g729| No Tx: ACK 192.168.1.13 1010 041c9a77455 00102/0 0x280100 (g729| No Tx: ACK 81.201.84.45 3473290576 PUM273-UMU5 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 2706513184 ISB67X-ZJQN 00101/00102 0x100 (g729) No Rx: ACK 81.201.84.45 4023308836 G7JP5O-AA4J 00101/00102 0x100 (g729) No Rx: ACK 192.168.1.13 1016 0758ea9a349 00102/0 0x0 (nothing) No (d) Tx: ACK 122.169.113.145 1006 379b29497d0 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 1006 3a4fc558695 00102/0 0x0 (nothing) No Init: NOTIFY 122.169.113.145 1006 7678828b011 00102/0 0x0 (nothing) No Init: NOTIFY 13 active SIP channels The last 3 rows have been there since past 6 days. There is no user 1006, logged into the system... I have 2 questions: (1) Where does Trixbox store this information (2) How can I periodically remove these records Thx in advance. Sanjay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peers Listed in sip show channels
do you have that user 1006 defined by IP ? *I have a user 1006. Its not defined by IP. * does it have mailbox= also defined ? *Yes. 1006 has a Mail box*. my wild guess is that there's unchecked voicemail and asterisk tries to initialize sending NOTIFY MWI messages *I will delete all messages from the Mailbox and see if 1006 is removed from the listing. * you can't remove these messages they remove themselves after some timeout *Any idea where there are 3 rows with 1006*? Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peers Listed in sip show channels
my wild guess is that there's unchecked voicemail and asterisk tries to initialize sending NOTIFY MWI messages *I will delete all messages from the Mailbox and see if 1006 is removed from the listing.* Just checked, no messages in 1006. Any other reasons! Thx Sanjay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callfile to auto-answering extension
I have a SPA742, which can autoanswer a call In the dialplan, I have this: exten = 28,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = 28,2,dial(SIP/36) Now I want some external event initiate a call to that phone and play a message. I have been thinking of dialfiles, but I believe there is a problem: Dialfiles call a channel, and then executes the dialplan. I need to SIPAddHeader first, then make the call. Or am I missing something obvious? Can I, via a callfile, or command-line parameters to Asterisk start a dialplan-script? eg asterisk -someflag execute callalert then in dialplan [callalert] exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = s,2,dial(SIP/36) exten = s,3,Playback(firealert) exten = s,4,Hangup Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: New in asterisk
With best regards Abdul Ahad Anwer Khan, M.Sc(CME, in progress) University of Applied Sciences Offenburg Germany Phone:+497814748226 Mobile:+4917623468462 From: abdulahadan...@hotmail.com To: asterisk-users-boun...@lists.digium.com Subject: New in asterisk Date: Sun, 27 Sep 2009 14:50:59 +0600 Hello All I am a student and doing my thesis which is related to asterisk. I am new in this field and hence facing a little bit problem. I have to work with AMI to do the call generation. I have two sip soft clients '6010' and '6011'. The asterisk I am working with is trixbox 2.6.2.3. To originate the call between the two softphones I have tried to use the following set of commands C:\telnet 192.168.0.72 5038 Asterisk Call Manager/1.1 Action: login Username: manager Secret: password Response: Success Message: Authentication accepted Action: Originate Channle: SIP/6010 Exten: 6011 Priority: 1 Timeout: 6 Context: default Response: Error Message: Premission denied Please let me know the remedy of this problem if it is possible?? or how could I acheive a calling mechanism between two softphones using AMI Waiting for the replies With best regards Abdul Ahad Anwer Khan See all the ways you can stay connected to friends and family _ Windows Live™: Keep your life in sync. Check it out! http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium fax: failed to queue document
Martin wrote:
u don't change the ${uniquefile} for the second System/Originate
try to add a string to the ${uniquefile} ...
eg
${uniquefile}0
Martin
But I generate another unique file in [fax-tx] just before I try to send
the confirm.
Here's the first call:
-- Executing [s...@outbound-fax:2] System(Console/dsp, env echo
-e Channel:DAHDI/g0/abbbccc\\nContext:fax-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254021344.0) in
new stack
Here's the second:
-- Executing [...@fax-tx:9] System(DAHDI/1-1, env echo -e
Channel:DAHDI/g0/abbbccc\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254021349.10)
in new stack
As you can see, I also tried adding the 0 string :)
Also here's tiffinfo for the file that's not queued:
[Sep 26 23:16:38] ERROR[18841]: res_fax_digium.c:1761 dgm_fax_start: fax
handle: 0 failed to queue document
'/var/spool/asterisk/fax-tx-status-20090926_2316.tif'
tiffinfo /var/spool/asterisk/fax-tx-status-20090926_2316.tif
TIFF Directory at offset 0x822 (2082)
Image Width: 600 Image Length: 200
Bits/Sample: 1
Compression Scheme: CCITT Group 3
Photometric Interpretation: min-is-black
FillOrder: msb-to-lsb
Orientation: row 0 top, col 0 lhs
Samples/Pixel: 1
Rows/Strip: 109
Planar Configuration: single image plane
DocumentName: Standard Input
ImageDescription: converted PNM file
sean
On Sat, Sep 26, 2009 at 8:05 PM, sean darcy seandar...@gmail.com wrote:
In my quest to actually send a fax, I'm now stuck trying to send the
confirm.
First I send the fax:
-- Executing [s...@outbound-fax:2] System(Console/dsp, env echo
-e Channel:DAHDI/g0/12036378447\\nContext:fax-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254012878.0) in
new stack
-- Auto fallthrough, channel 'Console/dsp' status is 'UNKNOWN'
Hangup on console
-- Attempting call on DAHDI/g0/12036378447 for s...@fax-tx:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
Channel DAHDI/1-1 was answered.
-- Executing [...@fax-tx:1] SendFAX(DAHDI/1-1,
/var/spool/asterisk/fax/20090922_1301.tif) in new stack
-- Channel 'DAHDI/1-1' sending fax
'/var/spool/asterisk/fax/20090922_1301.tif'
-- Channel 'DAHDI/1-1' fax session '0' started
.
And that works.
Then I try to send the confirm:
'h' =1. Set(RID=${FAXOPT(remotestationid)})
[pbx_config]
2.
Set(DateTime=${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
[pbx_config]
3.
Set(GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-${DateTime})
[pbx_config]
4. System(env echo -e ${FAXOPT(pages)} Page Fax
sent to ${EXTEN}. Remote ID: ${RID} ${StatusFile}-l1) [pbx_config]
5. System(env echo -e Status: ${FAXOPT(status)}
${FAXOPT(statusstr)} ${StatusFile}-l2) [pbx_config]
6. System(convert -background white -fill black
-pointsize 18 text:${StatusFile}-l1 text:${StatusFile}-l2 -crop
600x100+1+1 -append ${StatusFile}.tif) [pbx_config]
7. Set(GLOBAL(StatusFile)=${StatusFile})
[pbx_config]
8.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
9. System(env echo -e
Channel:DAHDI/g0/abbbccc\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n ${UniqueFile}) [pbx_config]
But that fails:
-- Executing [...@fax-tx:1] Set(DAHDI/1-1, RID=bbb-ccc-) in
new stack
-- Executing [...@fax-tx:2] Set(DAHDI/1-1,
DateTime=20090926_2055) in new stack
-- Executing [...@fax-tx:3] Set(DAHDI/1-1,
GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-20090926_2055) in
new stack
== Setting global variable 'StatusFile' to
'/var/spool/asterisk/fax-tx-status-20090926_2055'
-- Executing [...@fax-tx:4] System(DAHDI/1-1, env echo -e 1 Page
Fax sent to h. Remote ID: bbb-ccc-
/var/spool/asterisk/fax-tx-status-20090926_2055-l1) in new stack
-- Executing [...@fax-tx:5] System(DAHDI/1-1, env echo -e Status:
SUCCESS FAX_SUCCESS
/var/spool/asterisk/fax-tx-status-20090926_2055-l2) in new stack
-- Executing [...@fax-tx:6] System(DAHDI/1-1, convert -background
white -fill black -pointsize 18
text:/var/spool/asterisk/fax-tx-status-20090926_2055-l1
text:/var/spool/asterisk/fax-tx-status-20090926_2055-l2 -crop
600x100+1+1 -append
/var/spool/asterisk/fax-tx-status-20090926_2055.tif) in new stack
-- Executing [...@fax-tx:7] Set(DAHDI/1-1,
GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-20090926_2055) in
new stack
== Setting global variable 'StatusFile' to
'/var/spool/asterisk/fax-tx-status-20090926_2055'
-- Executing [...@fax-tx:8] Set(DAHDI/1-1,
UniqueFile=/var/spool/asterisk/outgoing/call-1254012879.1) in new stack
-- Executing [...@fax-tx:9] System(DAHDI/1-1, env echo -e
Re: [asterisk-users] New thread - SIP over VPN
On Sat, 2009-09-26 at 22:47 -0700, Dave Platt wrote: Isn't an SSL based tunnel all TCP? There seems to be a good deal of feeling (and evidence) that trying to use TCP as the container for a tunnel is likely to cause more trouble than it solves. Yes, the TCP layer will make the tunnel reliable - but at the expense of adding unpredictable amounts of latency, due to TCP's built-in exponential-backoff retry timing. Things get *really* nasty if you try to wrap one TCP connection in another, because both connections will be independently retrying any lost or delayed packets - you'll end up retransmitting quite a bit more data than you would if you simply used TCP/IP (or TCP/IP wrapped in UDP/IP) and throughput will suffer. That is the main reason why the widespread of (TCP) SSH-tunnels is discouraged: as you get an TCP-protocol encapsulated in another TCP-layer. Missing frames will be corrected by the outermost TCP-protocal-suite, however as soon as you got a bad-connection (Often wifi) and are confronted with timeouts, re-transmissions will on make things worse. and end-up with a snowball-effect. So i would opt for ipsec-tunnel or openvpn with UDP. If you have a rock-solid connection you could even use an openSSH-vpn tunnel. hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callfile to auto-answering extension
2009/9/27 Leif Neland le...@neland.dk: Can I, via a callfile, or command-line parameters to Asterisk start a dialplan-script? eg asterisk -someflag execute callalert then in dialplan [callalert] exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = s,2,dial(SIP/36) exten = s,3,Playback(firealert) exten = s,4,Hangup ...sure, use Local channels. You can use Local/ext@context as the originating channel in a call file or AMI/CLI originate command. -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Know for how long an agent is talking?
On Sun, Sep 27, 2009 at 12:07 AM, Gabriel Ortiz Lour ortiz.ad...@gmail.com wrote: Hi, Is there a way to know for how long an agent is talking on the queue call? (without keeping a timer myself... just asking asterisk) Identify the channel at the CLI and then get its details via core show channel channel-spec. Asterisk will gladly give you lots of details regarding that channel, including the channel uptime. Does this answer your question or did I misunderstood it ? -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are phone registrations kept?
I've been willing to give such a solution a try but the lack of time has prevented it to date... Are you using realtime for your SIP peers/users ? Would the failover behaviour improve under such scenario ? (just a thought) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with Digium TDM400 card
I had a working Asterisk 1.4.24.1 installation on SUSE 9 Linux but SIP only. I then downloaded and installed latest Zaptel and could not get Zaptel working. So I downloaded Asterisk again and re-installed. But still problems: Here is my ztcfg output: asterisk:/etc/asterisk # ztcfg -v Notice: Configuration file is /etc/zaptel.conf line 4: Unable to read Zaptel version information. Zaptel Version: Unknown Echo Canceller: Unknown Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) My zaptel.conf file is in /etc/ fxsks=1-4 loadzone=uk defaultzone=uk I know config above is correct because used to work in an older Asterisk 1.2 installation. My zapata.conf file is in /etc/asterisk/ has this setting: signalling=fxs_ks What should I be looking at? Works ok for SIP but I want to get the analog card working. It is a TDM04B. Angus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callfile to auto-answering extension
Ex Vito skrev: 2009/9/27 Leif Neland le...@neland.dk: Can I, via a callfile, or command-line parameters to Asterisk start a dialplan-script? eg asterisk -someflag execute callalert then in dialplan [callalert] exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = s,2,dial(SIP/36) exten = s,3,Playback(firealert) exten = s,4,Hangup ...sure, use Local channels. You can use Local/ext@context as the originating channel in a call file or AMI/CLI originate command. -- Sorry, I'm a little rusty... What exactly do I write, If I want to use a CLI originate command, to execute the above callalert? Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Digium TDM400 card
On Sun, Sep 27, 2009 at 05:17:33PM +0100, Angus Asterisk wrote: I had a working Asterisk 1.4.24.1 installation on SUSE 9 Linux but SIP only. I then downloaded and installed latest Zaptel I'll note that for many practical purposes, latest zaptel is DAHDI. and could not get Zaptel working. So I downloaded Asterisk again and re-installed. But still problems: Here is my ztcfg output: asterisk:/etc/asterisk # ztcfg -v Notice: Configuration file is /etc/zaptel.conf line 4: Unable to read Zaptel version information. Zaptel Version: Unknown Echo Canceller: Unknown Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) ztcfg sent an ioctl that the kernel part of Zaptel didn't understand. This may be due to using a (very old?) version of zaptel.h to build ztcfg (or maybe way older modules and a newer ztcfg). A second option is mixing 64bit kernel with 32bit userspace. Also, what is the output of: uname -a which ztcfg file `which ztcfg` # let's hope this one exists: cat /sys/module/zaptel/version modinfo zaptel | grep ^version rpm -qa | grep zaptel -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Digium TDM400 card
At 09:17 AM 9/27/2009, you wrote: What should I be looking at? Works ok for SIP but I want to get the analog card working. It is a TDM04B. Have you tried running genzaptelconf or whatever it's called? Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is channel local what I need?
On 1.6.0.16-rc1:
I'm using app_fax.so to send a fax, and then send a confirm.
'send' = 1.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
2. System(env echo -e
Channel:DAHDI/g0/\\nContext:fax-tx\\nExtension: s\\nPriority:
1\\n ${UniqueFile}) [pbx_config]
[ Context 'fax-tx' created by 'pbx_config' ]
's' =1. SendFAX(${FaxFile}.tif)
[pbx_config]
'h' =1. Set(RID=${REMOTESTATIONID})
[pbx_config]
..
7. Set(GLOBAL(StatusFile)=${StatusFile})
[pbx_config]
8.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}0) [pbx_config]
9. System(env echo -e
Channel:DAHDI/g0/...\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n ${UniqueFile}) [pbx_config]
The first fax goes through, but the status fax dies:
-- Executing [...@fax-tx:9] System(DAHDI/1-1, env echo -e
Channel:DAHDI/g0/...\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254077933.390)
in new stack
-- Hungup 'DAHDI/1-1'
[Sep 27 14:59:24] NOTICE[8413]: pbx_spool.c:357 attempt_thread: Call
completed to DAHDI/g0/..
-- Attempting call on DAHDI/g0/. for s...@fax-confirm-tx:1
(Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
Channel DAHDI/1-1 was answered.
-- Executing [...@fax-confirm-tx:1] SendFAX(DAHDI/1-1,
/var/spool/asterisk/tmp/fax-tx-status-20090927_1459.tif) in new stack
[Sep 27 15:00:57] WARNING[8433]: app_fax.c:178 phase_e_handler: Error
transmitting fax. result=2: Timed out waiting for initial communication.
I'm also the receiving side. The first fax rings and is answered. The
status fax never rings.
Now I think the problem is that I'm in channel DAHDI/1-1, but that
channel has been hungup.
So...I tried:
9. Dial(local/s...@transfer-to-confirm/n)[pbx_config]
[ Context 'transfer-to-confirm' created by 'pbx_config' ]
's' = 1. NoOp(YooHoo) [pbx_config]
2. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID})
[pbx_config]
3. NoOp(YooHoo2) [pbx_config]
4. System(env echo -e
Channel:DAHDI/g0/.\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n ${UniqueFile}) [pbx_config]
But that just dies:
-- Executing [...@fax-tx:9] Dial(DAHDI/1-1,
local/s...@transfer-to-confirm/n) in new stack
-- Called s...@transfer-to-confirm/n
== Spawn extension (fax-tx, h, 9) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
[Sep 27 15:17:41] NOTICE[8607]: pbx_spool.c:357 attempt_thread: Call
completed to DAHDI/g0/12036378447
-- Executing [...@transfer-to-confirm:1]
NoOp(Local/s...@transfer-to-confirm-fdda;2, YooHoo) in new stack
And that's it. It never executes priority 2, or anything else.
sean
___
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[asterisk-users] channel.c:780 channel_find_locked: Avoided deadlock
Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to the carries... I can call using eyebeam directly and I can start the call. Asterisk is running ok, but when I tried to do a call with vicidial my phone ring but when I pick up the phone, the CLI show: WARNING[16629]: channel.c:780 channel_find_locked: Avoided deadlock. And the call is active, I can talk and listen to my partner :s But reseraching a lot in google, that warning sometimes doesnt afect the call. in some cases the call is going and I maybe that messages IS AFTER hung up. What can be wrong here ? :s This is part of /var/log/asterisk/messages: DEBUG[9326] chan_iax2.c: Immediately destroying 1, having received hangup Sep 27 16:20:36 VERBOSE[10049] logger.c: == Spawn extension (default, 8600051, 1) exited non-zero on 'IAX2/172.22.0.5:4569-1' Sep 27 16:20:36 VERBOSE[10049] logger.c: -- Executing DeadAGI(IAX2/172.22.0.5:4569-1, agi://127.0.0.1:4577/call_log) in new stack Sep 27 16:20:36 VERBOSE[10049] logger.c: -- AGI Script agi:// 127.0.0.1:4577/call_log completed, returning 0 Sep 27 16:20:36 VERBOSE[10049] logger.c: -- Executing DeadAGI(IAX2/172.22.0.5:4569-1, agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-16---)) in new stack Sep 27 16:20:36 VERBOSE[10049] logger.c: -- AGI Script agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-16---) completed, returning 0 Sep 27 16:20:36 DEBUG[10049] chan_iax2.c: We're hanging up IAX2/172.22.0.5:4569-1 now... Sep 27 16:20:36 DEBUG[10049] chan_iax2.c: Really destroying IAX2/172.22.0.5:4569-1 now... Sep 27 16:20:36 VERBOSE[10049] logger.c: -- Hungup 'IAX2/172.22.0.5:4569 -1' and this is the log CLI: Spawn extension (default, 8365, 4) exited non-zero on 'IAX2/172.22.0.5:4569 -4' -- Executing DeadAGI(IAX2/172.22.0.5:4569-4, agi:// 127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing DeadAGI(IAX2/172.22.0.5:4569-4, agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0---)) in new stack -- AGI Script agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-0---) completed, returning 0 -- Hungup 'IAX2/172.22.0.5:4569-4' In the other asterisk server (B whiche connect with the carries) I get this: Executing Dial(IAX2/vicidial-9945, SIP/17863126...@nettthone) in new stack -- Called 17863126...@nehone -- Accepting AUTHENTICATED call from 192.168.0.144: requested format = slin, requested prefs = (gsm|ulaw), actual format = ulaw, host prefs = (ulaw|alaw|gsm|ilbc), priority = mine -- Executing Dial(IAX2/vicidial-9585, SIP/17863126...@netttphone) in new stack -- Called 17863126...@nehone -- SIP/nettthone-08873100 is making progress passing it to IAX2/vicidial-9366 -- SIP/netttphone-088b37a0 is making progress passing it to IAX2/vicidial-9585 -- SIP/netttphone-0889ab98 is making progress passing it to IAX2/vicidial-9945 -- SIP/netttphone-0889ab98 answered IAX2/vicidial-9945 -- SIP/netttphone-088b37a0 answered IAX2/vicidial-9585 == Spawn extension (vicidial, 017863126954, 1) exited non-zero on 'IAX2/vicidial-9945' -- Hungup 'IAX2/vicidial-9945' -- --Michael Mendoza-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with incoming caller-ID to NEC SV8300 with QSIG
I'm using QSIG between an NEC SV8300 and Asterisk (after giving up with CCIS). Things work pretty well with the exception of issues on stations on the SV8300. When I call from Asterisk to a SV8300 station and I send my extension as the caller ID number, it shows up on the SV8300 as OPERATOR. I've tried a few different TON/NPI values with no difference. If I set callerid to a ten-digit number, it comes to the SV8300 with the last digit truncated. The name doesn't come over at all. When I call an Asterisk extension on an SV8300 station, the display doesn't show the station on the other side. When I look at PRI debug information, I think I see the SV8300 sending such information to Asterisk, but it seems to be discarded. I don't see Asterisk sending any such. libpri also complains about an unknown IE50. Before I start digging in to the 600+ pages of Q.931 and Q.932 specifications and the 4000+ lines of q931.c, I thought I'd ask if anybody else has run into these issues. Using the SV8300 to make an external call seems fine and everything from the SV8300 to Asterisk is fine when it comes to caller ID (number and name). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI congestion problem
I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1, DAHDI/g0/9239220,300,) in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the group is valid: dahdi show channels group g0 Chan Extension Context Language MOH InterpretBlocked State 4from-zaptel en default In Service The only Warning or Error I see is when asterisk first starts a new call. logger.c: -- Starting simple switch on 'DAHDI/1-1' [Sep 27 15:55:50] WARNING[4199] chan_dahdi.c: Unable to enable echo cancellation on channel 1 (No such device) On my TDM400P card, channel 1 is my analog phone, 2 my fax, and 4 the POTS line. More config files etc below. Any ideas? Thanks, Andy /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 10 22:20:05 2009 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) fxols=1 #echocanceller=mg2,1 fxols=2 #echocanceller=mg2,2 # channel 3, WCTDM/4/2, no module. fxsks=4 echocanceller=mg2,4 # Global data loadzone= us defaultzone = us dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) dahdi show channel 4 Channel: 4I File Descriptor: 20 Span: 1*CLI Extension: Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0I Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: yes Busy Count: 6 Busy Pattern: 0,0 TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI congestion problem
Andy Howell wrote: I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1, DAHDI/g0/9239220,300,) in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the group is valid: dahdi show channels group g0 Chan Extension Context Language MOH InterpretBlocked State 4from-zaptel en default In Service The only Warning or Error I see is when asterisk first starts a new call. logger.c: -- Starting simple switch on 'DAHDI/1-1' [Sep 27 15:55:50] WARNING[4199] chan_dahdi.c: Unable to enable echo cancellation on channel 1 (No such device) On my TDM400P card, channel 1 is my analog phone, 2 my fax, and 4 the POTS line. More config files etc below. Any ideas? Thanks, Andy /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 10 22:20:05 2009 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) fxols=1 #echocanceller=mg2,1 fxols=2 #echocanceller=mg2,2 # channel 3, WCTDM/4/2, no module. fxsks=4 echocanceller=mg2,4 # Global data loadzone= us defaultzone = us dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) dahdi show channel 4 Channel: 4I File Descriptor: 20 Span: 1*CLI Extension: Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0I Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: yes Busy Count: 6 Busy Pattern: 0,0 TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry to reply to my own post. I found that if I receive a call, I can then make outgoing calls until I reboot again. There must be something on my config that doesn't fully initialize the card. Once I've received a call, I can restart asterisk and it still works. Weird. Regards, Andy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe Hints
I've got hints setup for my MeetMe conferences like so:
exten = _60X,hint,MeetMe:${EXTEN}
and they show up in core show hints like so
6...@dialtone: MeetMe:600State:Unavailable
Watchers 1
_...@dialtone: MeetMe:${EXTEN} State:Unavailable
Watchers 0
I'm wondering why they're Unavailable instead of Idle. They go to
State:InUse when active but usually return to Unavailable when the
conference ends. Occasionally they end up in InUse but not
consistently.
Anybody know why?
Paul
--
Paul Dugas -- Computer Engineer -- Dugas Enterprises, LLC
522 Black Canyon Park, Canton GA 30114 USA
p...@dugasenterprises.com -- +1.404.932.1355
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[asterisk-users] DAHDI Question/Choppy Sound
Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring dummy DAHDI for Asterisk 1.6? Thanks in advance for any suggestion(s). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
