[asterisk-users] pjsip extensions rings but call drop on answer
serializer pjsip/outsess/4053-0077 associated with dialog dlg0x7f0578069118 [Jun 8 12:28:09] DEBUG[4181] res_pjsip_transport_websocket.c: Response msg 180/INVITE/cseq=30101 (rdata0x7f057808bd18) re-writing Contact URI from 10.215.144.48:64842;transport=ws to 10.215.144.48:64842;transport=ws [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: Function session_inv_on_state_changed called on event TSX_STATE [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: The state change pertains to the endpoint '4053(PJSIP/4053-0002)' [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f0570019af8) [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: There is no transaction involved in this state change [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: The current inv state is EARLY [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: Source of transaction state change is RX_MSG [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: Received response [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: Response is 180 Ringing [Jun 8 12:28:09] DEBUG[4164] devicestate.c: No provider found, checking channel drivers for PJSIP - 4053 [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: Function session_inv_on_tsx_state_changed called on event TSX_STATE [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: The state change pertains to the endpoint '4053(PJSIP/4053-0002)' [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: The inv session still has an invite_tsx (0x7f0570019af8) [Jun 8 12:28:09] DEBUG[4181] res_pjsip_session.c: The UAC INVITE transaction involved in this state change is 0x7f0570019af8 There is a firewall in the middle, but all ports and protocols are allowed. Any ideas? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk video streaming
Hi, I'd like an Asterisk SIP client with videosupport=yes to be able to dial into an IVR which would allow the user (after asking a few questions - like authentication, etc) to stream it's audio and video (mic + webcam) as an HTTP stream (say, a flash stream as it's the most common format for now or webm, vp8, etc.). How can I do this? The setup would be something like this: SIP client with webcam---Asterisk Server1 SIP IVR---video and audio conversion + streaming as HTTP---Internet clients (flash players or built-in modern browser players) From extensions.conf (within my IVR) can I call an AGI script that will then redirect both audio and video to an external application on the Asterisk server such as VideoLAN's VLC? I don't know if the external app can take the videoaudio from Asterisk as INPUT, transcode it and stream it as, eg., an FLV via HTTP. There are lots of streaming solutions out there (red5, vlc, ffmpeg) but I'd really like to know if someone here already has experience connecting Asterisk to one of these solutions and how. Or can Asterisk 11 already do the HTTP streaming part on its own? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple sipusers tables
Hi, I have 2 databases DBNAME1 and DBNAME2. In each database I have a table called sipusers (so DBNAME1.sipusers and DBNAME2.sipusers). Can I use both sipusers tables in Asterisk RealTime? Something like this: /etc/asterisk/extconfig.conf: [settings] sipusers = odbc,DBNAME1,sipusers sippeers = odbc,DBNAME1,sipusers sipusers = odbc,DBNAME2,sipusers sippeers = odbc,DBNAME2,sipusers If Asterisk 11 doesn't support this right now, will it in the future? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple sipusers tables
Right, thanks! --- On Tue, 3/5/13, Gertjan Baarda gertjan.baa...@gmail.com wrote: Maybe you can workaround it by creating a view in SQL?-- Gertjan On Tue, Mar 5, 2013 at 2:10 PM, Vieri rentor...@yahoo.com wrote: Hi, I have 2 databases DBNAME1 and DBNAME2. In each database I have a table called sipusers (so DBNAME1.sipusers and DBNAME2.sipusers). Can I use both sipusers tables in Asterisk RealTime? Something like this: /etc/asterisk/extconfig.conf: [settings] sipusers = odbc,DBNAME1,sipusers sippeers = odbc,DBNAME1,sipusers sipusers = odbc,DBNAME2,sipusers sippeers = odbc,DBNAME2,sipusers If Asterisk 11 doesn't support this right now, will it in the future? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Macro are evil :-)
This link may be a bit too old:
https://issues.asterisk.org/jira/browse/ASTERISK-9518
so maybe using MACRO_EXTEN won't work with macros in AEL.
Haven't tried that.
Vieri
--- On Sun, 2/24/13, Mitul Limbani mi...@enterux.in wrote:
Hi,
You might want to use ${MACRO_EXTEN} variable inside to preserve exten variable
of the original dialplan exten variable.
Mitul
On Feb 24, 2013 4:04 PM, Leandro Dardini ldard...@gmail.com wrote:
I just discover an hidden problem with AEL macro I want to have your
feedback. If you use a macro to dial out, like dialout(${EXTEN}), the leg
extension will became s and if it happens you transfer the call, that
will be the callerid appearing on the other phone display.
I am just rewriting all the dialplan getting rid of the macro and using gosub,
even if asterisk is complaining about application call to gosub affects flow
of control, and needs to be re-written using AEL if, while, goto, etc. keywords
instead!, but I am not seeing any other way...
Leandro
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[asterisk-users] asterisk 11 and H323
Hi, Could anyone please point me to a comprehensive how-to for H323 support in Asterisk 11? I'd like to connect machines that only support H323 and Asterisk 11. I've read the h323.conf file but I'd like to see more example setups. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11 and H323
Is chan_ooh323 broken in Asterisk 11? --- On Tue, 2/5/13, Vieri rentor...@yahoo.com wrote: Hi, Could anyone please point me to a comprehensive how-to for H323 support in Asterisk 11? I'd like to connect machines that only support H323 and Asterisk 11. I've read the h323.conf file but I'd like to see more example setups. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BRI D-channel goes up and down
Hi, I have a B410P card with span ports set up as span=3,1,0,CCS,AMI span=4,2,0,CCS,AMI span=5,3,0,CCS,AMI signalling = bri_cpe switchtype = euroisdn layer1_presence = ignore However, I keep getting these messages over and over again: [Dec 14 18:53:14] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 3 up == Primary D-Channel on span 4 up == Primary D-Channel on span 5 down [Dec 14 18:53:25] WARNING[22478]: sig_pri.c:1150 pri_find_dchan: Span 5: D-channel is down! == Primary D-Channel on span 5 up == Primary D-Channel on span 4 down [Dec 14 18:53:30] WARNING[22477]: sig_pri.c:1150 pri_find_dchan: Span 4: D-channel is down! == Primary D-Channel on span 3 down [Dec 14 18:53:30] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 4 up == Primary D-Channel on span 3 up It seems I can dial out and in but I'm afraid I may be losing some calls if they happen to dial in/out right when a span goes down. libpri-1.4.13 dahdi-2.6.1 asterisk-11.0.1 Any suggestions? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI can receive calls but cannot dial out
--- On Fri, 12/7/12, Steve Totaro stot...@totarotechnologies.com wrote: Why don't your span numbers match? 1-4 but you have 3-6 in your .conf. What do you mean? I have the following: span=3,1,0,CCS,AMI span=4,2,0,CCS,AMI span=5,3,0,CCS,AMI span=6,4,0,CCS,AMI The first parameter is the port number (3-6). The second parameter is Timing (1-4). Is it mandatory to begin the port numbering with 1? Or does it simply have to be sequential? Anyway, I set the span port numbers from 3 to 6 because I based myself on the output of dahdi_scan which was the following: # dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 00 Slot 04 basechan=1 totchans=4 irq=18 type=analog port=1,FXO port=2,FXO port=3,FXO port=4,FXO [2] active=yes alarms=OK description=Wildcard TDM2400P name=WCTDM/0 manufacturer=Digium devicetype=Wildcard TDM2400P location=PCI Bus 00 Slot 05 basechan=5 totchans=24 irq=20 type=analog port=5,FXO port=6,FXO port=7,FXO port=8,FXO port=9,FXO port=10,FXO port=11,FXO port=12,FXO port=13,none port=14,none port=15,none port=16,none port=17,none port=18,none port=19,none port=20,none port=21,none port=22,none port=23,none port=24,none port=25,none port=26,none port=27,none port=28,none [3] active=yes alarms=OK description=B4XXP (PCI) Card 0 Span 1 name=B4/0/1 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=29 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [4] active=yes alarms=OK description=B4XXP (PCI) Card 0 Span 2 name=B4/0/2 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=32 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [5] active=yes alarms=OK description=B4XXP (PCI) Card 0 Span 3 name=B4/0/3 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=35 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS [6] active=yes alarms=RED description=B4XXP (PCI) Card 0 Span 4 name=B4/0/4 manufacturer=Digium devicetype=Wildcard B410P location=PCI Bus 00 Slot 06 basechan=38 totchans=3 irq=23 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI,HDB3 framing_opts=ESF,D4,CCS,CRC4 coding=AMI framing=CCS I assumed I should use as port numbers the values within square brackets above. Still, I'm wondering why outgoing calls don't work (dial/g2 in my example) if I disconnect the cable from: span=3,1,0,CCS,AMI and leave all the others connected. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI can receive calls but cannot dial out
--- On Fri, 12/7/12, Alex Kauffmann akauf...@prodigy.net.mx wrote: From: Alex Kauffmann akauf...@prodigy.net.mx Subject: Re: [asterisk-users] PRI can receive calls but cannot dial out To: asterisk-users@lists.digium.com Date: Friday, December 7, 2012, 11:37 AM On 12/7/2012 6:23 AM, Vieri wrote: Am 05.12.2012 08:48, schrieb Vieri: Hi, I'm trying to call out from a SIP extension to an outbound destination via a PRI E1 (Digium B410P). Please take a look at the PRI debug below. # cat /etc/dahdi/system.conf # Digium Wildcard TDM400P REV I (WCTDM/4) fxsks=1 echocanceller=oslec,1 fxsks=2 echocanceller=oslec,2 fxsks=3 echocanceller=oslec,3 fxsks=4 echocanceller=oslec,4 # Digium Wildcard TDM2400P (WCTDM/0) fxsks=5 echocanceller=oslec,5 fxsks=6 echocanceller=oslec,6 fxsks=7 echocanceller=oslec,7 fxsks=8 echocanceller=oslec,8 fxsks=9 echocanceller=oslec,9 fxsks=10 echocanceller=oslec,10 fxsks=11 echocanceller=oslec,11 fxsks=12 echocanceller=oslec,12 # Digium Wildcard B410P (B4/0/1) span=3,1,0,CCS,AMI bchan=29-30 hardhdlc=31 echocanceller=oslec,29-30 # Digium Wildcard B410P (B4/0/2) span=4,2,0,CCS,AMI bchan=32-33 hardhdlc=34 echocanceller=oslec,32-33 # Digium Wildcard B410P (B4/0/3) span=5,3,0,CCS,AMI bchan=35-36 hardhdlc=37 echocanceller=oslec,35-36 # Digium Wildcard B410P (B4/0/4) span=6,4,0,CCS,AMI bchan=38-39 hardhdlc=40 echocanceller=oslec,38-39 # lsmod | grep wcb4xxp wcb4xxp 66250 12 dahdi 169899 65 dahdi_echocan_oslec,wcb4xxp,wctdm24xxp,dahdi_voicebus,wctdm # cat chan_dahdi.conf [trunkgroups] [channels] transfer = yes usecallerid = yes cidsignalling = dtmf callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes canpark = yes cancallforward = yes callreturn = yes callprogress = no overlapdial = yes echocancel = yes facilityenable = yes immediate = no busydetect = no ; Digium Wildcard TDM400P REV I (WCTDM/4) signalling = fxs_ks txgain = 1.0 rxgain = 14.0 group = 3 context = incoming-dahdi-3 faxdetect = incoming channel = 1,2,3,4 ; Digium Wildcard TDM2400P (WCTDM/0) group = 4 context = incoming-dahdi-4 faxdetect = incoming channel = 5,6,7,8,9,10,11,12 ; Digium Wildcard B410P (B4/0/1) signalling = bri_cpe switchtype = euroisdn rxgain = 2.0 group = 2 context = incoming-dahdi-2 faxdetect = incoming channel = 29-30 ; Digium Wildcard B410P (B4/0/2) channel = 32-33 ; Digium Wildcard B410P (B4/0/3) channel = 35-36 ; Digium Wildcard B410P (B4/0/4) channel = 38-39 --- # asterisk -rx dahdi show status Description Alarms IRQ bpviol CRC Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) Wildcard TDM2400P OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 1 RED 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 2 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 3 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 4 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) Note that I have 3 cables connected and 1 port is free (RED). --- in AEL dialplan, I run: Dial(DAHDI/g2/XX); in the *CLI I see the following: -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g2/XX -- Span 4: Channel 0/1 got hangup, cause 18 -- Hungup 'DAHDI/i4/XX-7' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4053-0089' status is 'CHANUNAVAIL' If I enable PRI debug: -- Executing [@company:1] Dial(SIP/4053-0001, DAHDI/g2/XX) in new stack PRI Span: 4 -- Making new call for cref 32772 -- Requested transfer capability: 0x00 - SPEECH PRI Span: 4 PRI Span: 4 DL-DATA request PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 TEI=0 Transmitting N(S)=6, window is open V(A)=6 K=1 PRI Span: 4 PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 [04 03 80 90 a3] PRI Span: 4 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 4 Ext: 1 Trans
Re: [asterisk-users] PRI can receive calls but cannot dial out
Am 05.12.2012 08:48, schrieb Vieri: Hi, I'm trying to call out from a SIP extension to an outbound destination via a PRI E1 (Digium B410P). Please take a look at the PRI debug below. # cat /etc/dahdi/system.conf # Digium Wildcard TDM400P REV I (WCTDM/4) fxsks=1 echocanceller=oslec,1 fxsks=2 echocanceller=oslec,2 fxsks=3 echocanceller=oslec,3 fxsks=4 echocanceller=oslec,4 # Digium Wildcard TDM2400P (WCTDM/0) fxsks=5 echocanceller=oslec,5 fxsks=6 echocanceller=oslec,6 fxsks=7 echocanceller=oslec,7 fxsks=8 echocanceller=oslec,8 fxsks=9 echocanceller=oslec,9 fxsks=10 echocanceller=oslec,10 fxsks=11 echocanceller=oslec,11 fxsks=12 echocanceller=oslec,12 # Digium Wildcard B410P (B4/0/1) span=3,1,0,CCS,AMI bchan=29-30 hardhdlc=31 echocanceller=oslec,29-30 # Digium Wildcard B410P (B4/0/2) span=4,2,0,CCS,AMI bchan=32-33 hardhdlc=34 echocanceller=oslec,32-33 # Digium Wildcard B410P (B4/0/3) span=5,3,0,CCS,AMI bchan=35-36 hardhdlc=37 echocanceller=oslec,35-36 # Digium Wildcard B410P (B4/0/4) span=6,4,0,CCS,AMI bchan=38-39 hardhdlc=40 echocanceller=oslec,38-39 # lsmod | grep wcb4xxp wcb4xxp 66250 12 dahdi 169899 65 dahdi_echocan_oslec,wcb4xxp,wctdm24xxp,dahdi_voicebus,wctdm # cat chan_dahdi.conf [trunkgroups] [channels] transfer = yes usecallerid = yes cidsignalling = dtmf callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes canpark = yes cancallforward = yes callreturn = yes callprogress = no overlapdial = yes echocancel = yes facilityenable = yes immediate = no busydetect = no ; Digium Wildcard TDM400P REV I (WCTDM/4) signalling = fxs_ks txgain = 1.0 rxgain = 14.0 group = 3 context = incoming-dahdi-3 faxdetect = incoming channel = 1,2,3,4 ; Digium Wildcard TDM2400P (WCTDM/0) group = 4 context = incoming-dahdi-4 faxdetect = incoming channel = 5,6,7,8,9,10,11,12 ; Digium Wildcard B410P (B4/0/1) signalling = bri_cpe switchtype = euroisdn rxgain = 2.0 group = 2 context = incoming-dahdi-2 faxdetect = incoming channel = 29-30 ; Digium Wildcard B410P (B4/0/2) channel = 32-33 ; Digium Wildcard B410P (B4/0/3) channel = 35-36 ; Digium Wildcard B410P (B4/0/4) channel = 38-39 --- # asterisk -rx dahdi show status Description Alarms IRQ bpviol CRC Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) Wildcard TDM2400P OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 1 RED 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 2 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 3 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 4 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) Note that I have 3 cables connected and 1 port is free (RED). --- in AEL dialplan, I run: Dial(DAHDI/g2/XX); in the *CLI I see the following: -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g2/XX -- Span 4: Channel 0/1 got hangup, cause 18 -- Hungup 'DAHDI/i4/XX-7' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4053-0089' status is 'CHANUNAVAIL' If I enable PRI debug: -- Executing [@company:1] Dial(SIP/4053-0001, DAHDI/g2/XX) in new stack PRI Span: 4 -- Making new call for cref 32772 -- Requested transfer capability: 0x00 - SPEECH PRI Span: 4 PRI Span: 4 DL-DATA request PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 TEI=0 Transmitting N(S)=6, window is open V(A)=6 K=1 PRI Span: 4 PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 [04 03 80 90 a3] PRI Span: 4 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 4 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 4 User information layer 1: A-Law (35) PRI Span: 4 [18 01 81] PRI Span: 4 Channel ID (len= 3) [ Ext: 1 IntID: Implicit BRI Spare: 0 Preferred Dchan: 0
[asterisk-users] PRI can receive calls but cannot dial out
Preferred Dchan: 0 PRI Span: 4ChanSel: B1 channel PRI Span: 4 ] PRI Span: 4 [6c 06 21 80 34 30 35 33] PRI Span: 4 Calling Party Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 4 Presentation: Presentation allowed, User-provided, not screened (0) '4053' ] PRI Span: 4 [70 0a 80 36 35 36 36 36 30 34 39 39] PRI Span: 4 Called Party Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'XX' ] PRI Span: 4 q931.c:6291 q931_setup: Call 32774 enters state 1 (Call Initiated). Hold state: Idle -- Called DAHDI/g2/XX PRI Span: 4 T303 timed out. cref:32774 PRI Span: 4 PRI Span: 4 DL-DATA request PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 6/0x6) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 TEI=0 Transmitting N(S)=11, window is open V(A)=11 K=1 PRI Span: 4 PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 6/0x6) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 [04 03 80 90 a3] PRI Span: 4 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 4 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 4 User information layer 1: A-Law (35) PRI Span: 4 [18 01 81] PRI Span: 4 Channel ID (len= 3) [ Ext: 1 IntID: Implicit BRI Spare: 0 Preferred Dchan: 0 PRI Span: 4ChanSel: B1 channel PRI Span: 4 ] PRI Span: 4 [6c 06 21 80 34 30 35 33] PRI Span: 4 Calling Party Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 4 Presentation: Presentation allowed, User-provided, not screened (0) '4053' ] PRI Span: 4 [70 0a 80 36 35 36 36 36 30 34 39 39] PRI Span: 4 Called Party Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'XX' ] PRI Span: 4 T303 timed out. cref:32774 PRI Span: 4 q931.c:6180 t303_expiry: Call 32774 enters state 0 (Null). Hold state: Idle PRI Span: 4 Fake clearing. cref:32774 PRI Span: 4 q931.c:9551 pri_internal_clear: alive 1, hangupack 1 Span 4: Processing event PRI_EVENT_HANGUP(6) -- Span 4: Channel 0/1 got hangup, cause 18 PRI Span: 4 q931.c:7092 q931_hangup: Hangup other cref:32774 PRI Span: 4 q931.c:6849 __q931_hangup: ourstate Null, peerstate Null, hold-state Idle PRI Span: 4 Destroying call 0xb85c61d0, ourstate Null, peerstate Null, hold-state Idle -- Hungup 'DAHDI/i4/XX-6' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4053-0003' status is 'CHANUNAVAIL' -- Executing [h@company:3] Hangup(SIP/4053-0003, ) in new stack == Spawn extension (company, h, 3) exited non-zero on 'SIP/4053-0003' Note that incoming calls via this PRI work correctly. Asterisk 11.0.1 latest libpri and dahdi. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pipe character in CDR user field
I'm trying to set a CDR userfield to a custom value. This value may contain a '|' but it's really just part of the value. However, Asterisk keeps warning me about the application delimiter not being a pipe. It's NOT an application delimiter (it's just part of a variable value) so I'm expecting Asterisk not to warn me about it. Is it expected behavior? Why? See the following log: SIP/4053-007bAGI Rx EXEC Set CDR(userfield)=|usr_r=vieri -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=|usr_r=vieri) [Nov 29 10:53:08] WARNING[4815]: pbx.c:1563 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Set(CDR(userfield)=|usr_r=vieri)) SIP/4053-007bAGI Tx 200 result=0 SIP/4053-007dAGI Rx EXEC Set CDR(userfield)=\|usr_r=vieri\ -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=|usr_r=vieri) [Nov 29 10:54:57] WARNING[4838]: pbx.c:1563 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Set(CDR(userfield)=|usr_r=vieri)) SIP/4053-007dAGI Tx 200 result=0 Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI exec command
Hi,
I'm trying to understand how AGI works.
I'm using php-agi library from sf.net.
*CLI agi show commands
Yes exec Executes a given Application
*CLI core show application set
My PHP-AGI script contains:
$AGI-exec(Set, CUSTOM_VAR=2);
$AGI-exec(NoOp, \DEBUG - ${CUSTOM_VAR}\);
An AGI debug from *CLI shows:
SIP/4053-004dAGI Rx EXEC Set CUSTOM_VAR=2
-- AGI Script Executing Application: (Set) Options: (CUSTOM_VAR=2)
SIP/4053-004dAGI Tx 200 result=0
SIP/4053-004dAGI Rx EXEC NoOp DEBUG -
-- AGI Script Executing Application: (NoOp) Options: (DEBUG - )
Why isn't CUSTOM_VAR set?
I know I could use agi command set variable but I'd like to know why the
above code doesn't seem to work.
Also, there's no AGI-specific command for NoCDR(). So I did something like this:
$AGI-exec(NoCDR, );
but the CDR was written to cdr-csv/Master.csv so I'm assuming I'm doing
something wrong with the agi exec command.
Any ideas?
Thanks,
Vieri
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Re: [asterisk-users] AGI exec command
Never mind. Figured it out. Sorry for the noise. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] catch-all extension in context
Hi,
Suppose I have the following in my AEL dialplan:
context incoming-1 {
_. = {
Set(GROUP()=1);
goto incoming|${EXTEN}|1;
}
};
context incoming-2 {
_. = {
Set(GROUP()=2);
goto incoming|${EXTEN}|1;
}
};
context incoming {
fax = {
Do stuff for incoming fax...
}
_. = {
Do stuff for incoming voice call...
}
};
faxdetection is activated.
I'm expecting 'incoming-1' and 'incoming-2' to goto 'incoming' EVEN if Asterisk
detects the call as being a fax BEFORE going to 'incoming'.
Is that correct? (ie. _. also matches 'fax')
Thanks,
Vieri
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[asterisk-users] realtime field names
Hi An Asterisk queue uses field names / config variables such as: announce-holdtime However, documentation regarding realtime is very unclear. voip-info.org suggests to use announce_holdtime. Is this correct? What about monitor-type? Should it be underscored too (monitor_type)? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime field names
--- On Fri, 10/5/12, Vieri rentor...@yahoo.com wrote: An Asterisk queue uses field names / config variables such as: announce-holdtime However, documentation regarding realtime is very unclear. voip-info.org suggests to use announce_holdtime. Is this correct? What about monitor-type? Should it be underscored too (monitor_type)? It seems that I can use underscores or dashes indistinctly. Is that true for all fields/tables? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RealTime table fields ordering
Hi, According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip: Quote: If you place ipaddr before host (in the case of dynamic), you will never load the public IP address of your sip device, as it will be overwritten when host is encountered. UnQuote. From the latest Asterisk source tarball, the 'contrib' directory contains several realtime MySQL table definitions. The sippeers table has column 'ipaddr' before column 'host'. Also, 'permit' comes before 'deny'. Same for allow/disallow. Shouldn't the correct RealTime column/field order be: deny, permit and disallow, allow and host, ipaddr? As a side note, the iaxfriends RealTime MySQL table definition in the 'contrib' directory lacks the deny/permit fields which are quite important. However, the iaxfriends table does have the 'ipaddr' field after the 'host' field and the 'allow' field after 'disallow'. Furthermore, the asterisk.org wiki at: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure shows the same disorder in deny/permit, allow/disallow and host/ipaddr (MySQL example for RealTime). So it seems that the contrib directory and the asterisk.org wiki are inconsistent and incomplete. Of course I understand that these are 'contributed' files but they should be proof-read by the Digium devs before packing them up into the official source tarball. Or am I wrong about my observations concerning field order and field omissions? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RealTime table fields ordering
--- On Fri, 9/28/12, Hans Witvliet aster...@a-domani.nl wrote: how about the line: `ipaddr` varchar(15) DEFAULT NULL, Wonder how they try to squeeze an IPv6 address in it... should be: `ipaddr` varchar(50) DEFAULT NULL, I think `ipaddr` varchar(45) DEFAULT NULL, should be enough. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] context local: unexpected KW _LOCAL
Hi,
Is the local context a reserved word in extensions.ael?
If so, what is it used for?
Can I define 'context local {};' somehow?
This is the error I'm getting:
ERROR[24659] ael.y: File: /etc/asterisk/extensions.ael, Line 67, Cols:
9-13: Error: syntax error, unexpected KW
_LOCAL, expecting 'default' or word
I don't mind using a different context name but would simply like to know why
this error shows up.
Thanks,
Vieri
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[asterisk-users] undefined symbols
Hi, I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings: loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister loader.c: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 loader.c: Error loading module 'app_voicemail.so': /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: ast_smdi_mwi_message_destroy loader.c: Error loading module 'res_fax_spandsp.so': /usr/lib/asterisk/modules/res_fax_spandsp.so: undefined symbol: ast_fax_tech_register loader.c: Error loading module 'res_agi.so': /usr/lib/asterisk/modules/res_agi.so: undefined symbol: ast_speech_start loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client loader.c: Error loading module 'chan_mgcp.so': /usr/lib/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc loader.c: Error loading module 'cdr_adaptive_odbc.so': /usr/lib/asterisk/modules/cdr_adaptive_odbc.so: undefined symbol: SQLFetch loader.c: Error loading module 'cdr_odbc.so': /usr/lib/asterisk/modules/cdr_odbc.so: undefined symbol: SQLRowCount loader.c: Error loading module 'cel_odbc.so': /usr/lib/asterisk/modules/cel_odbc.so: undefined symbol Any ideas? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] undefined symbols
Never mind. In order to fix those undefined symbols at startup, I needed to preload the following modules: [modules] autoload=yes preload = res_ael_share.so preload = res_speech.so preload = res_agi.so preload = res_smdi.so preload = res_odbc.so preload = res_fax.so preload = res_pktccops.so preload = res_jabber.so Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] opus codec
Hi, Will Asterisk support the OPUS codec? http://opus-codec.org/ Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF digits falsely detected
Hi, I have a context that basically does: Wait(1) Background(message) WaitExten(10) _6XX,1,DoSomething The problem is that when I reach this context and press some digits (eg. 6566604) then I can see in the log that Asterisk reads 6655666. So it's actually reading the digits twice. How can I avoid this? Incoming channel type is ISDN (mISDN). Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk queue: announce in more than one language or announceoverride
Hi, I'd like a single queue to announce the caller's position, etc., in more than one language without user interaction. ie. announce position in English then in French then in Spanish Is this possible (without ivr)? Can anyone please give me a Queue cmd example with 'announceoverride'? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hints and server-side DND (do not disturb)
--- On Wed, 4/18/12, Warren Selby wcse...@selbytech.com wrote: exten = *280,n,Set(DEVICE_STATE(Custom:lamp)=BUSY) Thanks! So in short, it's all about DEVICE_STATE or DEVSTATE for * 1.4. I've just one last issue and was wondering how to run the following command on a remote Asterisk server: Set(DEVSTATE(Custom:mycustomstate)=BUSY) ie. how can I set a DEVICE STATE from one Asterisk server to another (for clustering purposes). Can I do it via AMI by running something like this? Setvar(DEVSTATE(Custom:mycustomstate)=BUSY) Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hints and server-side DND (do not disturb)
Hi, Currently I'm using hints to determine SIP presence. As I understand it, a SIP extension can be labeled as busy, ringing, etc, based on a channel status. So a channel MUST be present. If it isn't then the extension is considered to be available. If my statement is correct then is there a way to set the extesnion as busy even if there's no channel associated with this extension? eg. when an extension sets server-side DND (Do Not Disturb), it actually sets a boolean value in astdb. Whenever asterisk tries to route a call to this extension, it first checks this value. Obviously, there's no way I can use hints in this scenario, or is there? Is it possible to somehow create a dummy channel whenever an extension sets server-side DND (custom context) and delete it whenever it unsets it? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExtensionStatus event
Hi, I'm wondering if someone has already done a web application that queries 'ExtensionStatus' events. On my web site I have an extension listing. Next to each number I'd like to add an icon or something that shows the extension status. I'd like this status to be as real-time as possible. Being a web app, I was thinking of doing javascript JSON calls to Asterisk AJAM every x seconds. Has anyone done this already? (so I don't need to reinvent the wheel) Are there better approaches than querying for the ExtensionSatus for each extension on a web page listing? Asterisk and HTTP daemon are on different machines. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] red5sip SIP ua can't register
Hi, I'm trying to make red5sip (http://red5phone.googlecode.com/svn/) register to Asterisk 10. I get the following error: NOTICE[17909]: chan_sip.c:25741 handle_request_register: Registration from '8933 sip:8933@' failed for '127.0.0.1:5070' - Not a local domain sip.conf does not define any domain= or realm= values (defaults). red5sip's settings specify the asterisk realm. What could I have misconfigured? On the other hand, if I setup red5sip to register to another Asterisk 1.4 server with the same SIP user credentials, it succeeds. I can't seem to detect the relevant difference between my 1.4 and 10 installations. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk vs FreeSwitch vs Asterisk + OpenSIP
--- On Fri, 2/10/12, Leandro Dardini ldard...@gmail.com wrote: mysql multimaster replication and asterisk realtime. Just a word of caution: I've had terrible luck with MySQL NDB tables in a multimaster setup. I'm not a big expert but v.5.0 and 5.1 have given me lots of reliability issues (I lost table data several times). I'd like to try postgresql in a multimaster setup. Realtime with a clustered database is a nice idea but is it reliable? Any long-term success stories? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] distributed queue information over several Asterisk nodes
Is it possible to distribute QUEUE information among several Asterisk nodes in a multimaster or load balancing setup? I haven't tried this yet but if one uses realtime with a clustered multimaster database and the queue agents/members are fixed SIP channels (eg. SIP/100) then I guess that the Queue app will be able to contact the member no matter to which Asterisk node it registered. However, what happens if incoming calls enter more than one queue (a queue on any Asterisk node, as it would be expected in a fully load-balanced setup)? Let's say QUEUE1 on ASTNODE1 has 1 incoming call waiting to be picked up and a second call comes in but enters QUEUE1 on ASTNODE2 which was previously empty. So for example, how can the caller in QUEUE1 on ASTNODE2 be placed in position 2 instead of 1? In other words, can the same QUEUE work/collaborate over different Asterisk nodes? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP hardware phones
I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when sharing the network with a PC (because PC apps may require gigabit speed). The day will come when medium or low-budget hardphones will have integrated gigabit switches. But is it THAT expensive to put in 2 gigabit ports in a hardphone nowadays? Or is it just marketing? How much would it take for Digium to sell their D40 phones with gigabit ports? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP trunk audio bad but is OK again after SIP re-registration
Hi, When *ANY* SIP client (softphone, hardphone, ATA) registers to an Asterisk server on my LAN and the extension dials out through a remote SIP provider, the audio is fine for a while. It then degrades and starts to be cracky/jittery. The extension can call once and again and it will always be bad. The only way to somehow fix the audio problem is to unregister the local SIP extension/hardphone/softphone and register it back to the same Asterisk server. I repeated the test several times and it seems to be reproducible. It apparently has nothing to do with my SIP provider or my DSL connection or router. It doesn't even seem to be a network problem on my side. Curiously though, it only happens if dialing out through the SIP provider... I thought maybe the Asterisk server's system clock could be an issue but it doesn't seem to be skewing off too quickly. Also, this problem started showing up 2 weeks ago. Before that, we've been making a lot of calls through the provider without a glitch. Nothing has changed as far as hardware and software is concerned. What could I try? How can I debug this? Why is re-registering the SIP extension making a difference? Any clues? Asterisk 1.4 Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP hardware phones
Let me answer that, Carlos. A big hospital. These big infrastructures can be quite outdated and messy. Getting someone to cable old parts of the buildings can be very expensive. However, replacing just the backbone switches is something they can afford. And they don't need PoE, really. What kind of applications benefit from gigabit speed? Well, plenty, such as MDs having to view a whole bunch of x-ray images of several patients, as fast as possible. Believe me, doctors aren't patient and Gbps makes a big difference. So basically, that's your answer: these sites don't need PoE, just Gbps and can't afford cabling a huge old building. Now, they don't care for PoE on the hardphones either. So in these cases, I think it's clearly justifiable to have a low-budget Digium D40 or Grandstream GXP280 with a 2-NIC Gbps switch. Not a big deal anyway, because they can always add a mini 5 or 8-port gigiabit switch for around 20$ between the wall socket and the hardphone+PC, but that just adds another appliance to the doctor's office... --- On Wed, 2/8/12, Carlos Alvarez car...@televolve.com wrote: From: Carlos Alvarez car...@televolve.com Subject: Re: [asterisk-users] SIP hardware phones To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, February 8, 2012, 9:26 AM If the customer is so cheap that they won't properly build out the network, why would they have gigabit switches to the desktop which have a limited set of applications that actually benefit from it? Then there's PoE, which is expensive to start and very expensive with gigabit. So this mythical customer is too cheap to cable, but will buy a gigabit switch of dubious value, will they buy a PoE gigabit switch? If not, why not buy a value-priced PoE 100m switch which has a clear benefit instead of a low-end GB switch of dubious value? I just don't see the fit, and I'm guessing the vendors don't either. What is the exact network topology (brands/models) and applications that justify GB to the desktop, don't justify additional cabling, and how do you account for PoE in this environment? On Wed, Feb 8, 2012 at 7:13 AM, Vieri rentor...@yahoo.com wrote: --- On Wed, 2/8/12, Jason W. Parks jason.w.pa...@gmail.com wrote: From everything I've researched to date, my understanding is most locations have chosen to double their port density and continue to service the phone and computer on separate ports than to share a single line for both computer and phone. Reason primarily mentioned being troubleshooting concerns. If this is the case, the second port is not required, and become nothing but another gimmick to sell to you. Is this everyone else's experience as well? Well, at some locations, for technical and mostly political reasons, doubling port density so that the computer connects to a separate port is too costly, way over what a 60$ hardphone can cost (eg. Grandstream GXP285). I'd be glad to pay just a tad more for hundreds of basic hardphones, just as long as they can do gigabit. Vieri-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User hit f to disconnect call.
--- On Thu, 1/26/12, Kevin P. Fleming kpflem...@digium.com wrote: From: Kevin P. Fleming kpflem...@digium.com Subject: Re: [asterisk-users] User hit f to disconnect call. To: asterisk-users@lists.digium.com Date: Thursday, January 26, 2012, 10:58 AM On 01/26/2012 07:22 AM, Vieri wrote: Hi, I was receiving fax calls just fine until recently. I'm now having random disconnections. Faxes are received over an ISDN BRI line and Asterisk 1.4 detects it and sends it to a iaxmodem (exten 10025 below). All's apparently as expected except for the fact that the following message comes up in the Asterisk log: User hit f to disconnect call. The iaxmodem log also shows a premature hangup (see below). I did a test fax call but I certainly didn't press any key to abort the call. What does that message mean? Asterisk log (0X is destination, Y is sending fax machine): [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [fax@from-pstn-deviate-custom:12] Dial(mISDN/6-u22326, IAX2/10025/0971847022|20|d) in new stack [Jan 26 13:46:13] DEBUG[619] chan_iax2.c: prepending 8 to prefs [Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Call accepted by 127.0.0.1 (format alaw) [Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Format for call is alaw [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Called 10025/0X [Jan 26 13:46:13] VERBOSE[619] logger.c: -- IAX2/10025-3460 is ringing [Jan 26 13:46:13] VERBOSE[619] logger.c: -- User hit f to disconnect call. [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Hungup 'IAX2/10025-3460' [Jan 26 13:46:13] VERBOSE[619] logger.c: == Spawn extension (from-pstn-deviate-custom, f, 0) exited non-zero on 'mISDN/6-u22326' [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [h@from-pstn-deviate-custom:1] Macro(mISDN/6-u22326, hangupcall) in new stack 'f' is the fake DTMF control frame used inside Asterisk to indicate that a CNG tone was detected. Do you have 'faxdetect' enabled on the mISDN channel driver for that BRI? Even if you do, though, I don't know why receiving an 'f' would disconnect the call, unless you've provided the 'd' option to app_dial. Even if you did, app_dial should be smart enough to not treat 'f' as a DTMF key, but it's not (at least not in Asterisk 1.4, this may have changed in later versions). That could be it. misdn has fax detection for incoming. app_dial IS using the 'd' option. I will try not to use it. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] User hit f to disconnect call.
Hi, I was receiving fax calls just fine until recently. I'm now having random disconnections. Faxes are received over an ISDN BRI line and Asterisk 1.4 detects it and sends it to a iaxmodem (exten 10025 below). All's apparently as expected except for the fact that the following message comes up in the Asterisk log: User hit f to disconnect call. The iaxmodem log also shows a premature hangup (see below). I did a test fax call but I certainly didn't press any key to abort the call. What does that message mean? Asterisk log (0X is destination, Y is sending fax machine): [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [fax@from-pstn-deviate-custom:12] Dial(mISDN/6-u22326, IAX2/10025/0971847022|20|d) in new stack [Jan 26 13:46:13] DEBUG[619] chan_iax2.c: prepending 8 to prefs [Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Call accepted by 127.0.0.1 (format alaw) [Jan 26 13:46:13] VERBOSE[15361] logger.c: -- Format for call is alaw [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Called 10025/0X [Jan 26 13:46:13] VERBOSE[619] logger.c: -- IAX2/10025-3460 is ringing [Jan 26 13:46:13] VERBOSE[619] logger.c: -- User hit f to disconnect call. [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Hungup 'IAX2/10025-3460' [Jan 26 13:46:13] VERBOSE[619] logger.c: == Spawn extension (from-pstn-deviate-custom, f, 0) exited non-zero on 'mISDN/6-u22326' [Jan 26 13:46:13] VERBOSE[619] logger.c: -- Executing [h@from-pstn-deviate-custom:1] Macro(mISDN/6-u22326, hangupcall) in new stack iaxmodem log: [2012-01-26 13:46:13] Incoming call connected 0X, Y, (null). [2012-01-26 13:46:13] Answering [2012-01-26 13:46:13] Remote hangup. [2012-01-26 13:46:14] Hanging Up [2012-01-26 13:46:19] Hanging Up [2012-01-26 13:46:22] Taking receiver off-hook. [2012-01-26 13:46:22] Hanging Up Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP hardphone with dual gigabit ethernet ports
Hi, I'm looking for a SIP hardphone with 2 network interfaces at 1 Gbps. All the ones I've seen only have dual 10/100Mbps ethernet ports (eg. Grandstream products). Any suggestions? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi: Unknown symbol kasprintf
When I compile dahdi I see these warnings: WARNING: kasprintf [dahdi-linux-2.5.0.2/drivers/dahdi/wctdm24xxp/wctdm24xxp.ko] undefined! WARNING: kasprintf [dahdi-linux-2.5.0.2/drivers/dahdi/dahdi.ko] undefined! And modinfo dahdi shows that the driver was built for a 2.6.17 kernel, SMP mod_unload 586 4KSTACKS gcc-4.1 If I modprobe -a dahdi, I get the following in dmesg: dahdi: Unknown symbol kasprintf Could this be a gcc/glibc or kernel headers issue? Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi: Unknown symbol kasprintf
--- On Wed, 12/21/11, Russ Meyerriecks rmeyerrie...@digium.com wrote: When I compile dahdi I see these warnings: WARNING: kasprintf [dahdi-linux-2.5.0.2/drivers/dahdi/wctdm24xxp/wctdm24xxp.ko] undefined! WARNING: kasprintf [dahdi-linux-2.5.0.2/drivers/dahdi/dahdi.ko] undefined! And modinfo dahdi shows that the driver was built for a 2.6.17 kernel, SMP mod_unload 586 4KSTACKS gcc-4.1 What distro are you running? A somewhat outdated Gentoo box. I couldn't wait longer so I'm in the process of doing a new, clean system installation. Thanks anyway for replying. I just hope that dahdi and asterisk will compile and run fine with gcc 4.5 and kernel 3.0. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi: Unknown symbol kasprintf
--- On Wed, 12/21/11, Shaun Ruffell sruff...@digium.com wrote: From: Shaun Ruffell sruff...@digium.com Subject: Re: [asterisk-users] dahdi: Unknown symbol kasprintf To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, December 21, 2011, 12:08 PM On Wed, Dec 21, 2011 at 08:45:32AM -0800, Vieri wrote: --- On Wed, 12/21/11, Russ Meyerriecks rmeyerrie...@digium.com wrote: When I compile dahdi I see these warnings: WARNING: kasprintf [dahdi-linux-2.5.0.2/drivers/dahdi/wctdm24xxp/wctdm24xxp.ko] undefined! WARNING: kasprintf [dahdi-linux-2.5.0.2/drivers/dahdi/dahdi.ko] undefined! And modinfo dahdi shows that the driver was built for a 2.6.17 kernel, SMP mod_unload 586 4KSTACKS gcc-4.1 What distro are you running? A somewhat outdated Gentoo box. I couldn't wait longer so I'm in the process of doing a new, clean system installation. Thanks anyway for replying. I just hope that dahdi and asterisk will compile and run fine with gcc 4.5 and kernel 3.0. I know this is too late for you but... Looks like kasprintf was first added to the kernel in 2.6.18, not prior to 2.6.12 like DAHDI currently believes. The following command on a checkout of the current trunk of DAHDI should allow you to build against the 2.6.17 kernel. $ curl https://github.com/sruffell/dahdi-linux/commit/cbd536aea83.patch; | patch -p1 Thanks for the information. It will be useful for other systems I need to upgrade. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TE205P leds flash red on startup
Hi, I have a new Digium TE205P 2-span E1 card I just installed on a server. As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? I don't have another machine to test at the moment but would like to know what to expect. I have several single-span E1 cards and when the machine boots, their leds are off until the kernel module is loaded. What could be the problem with my TE205P? Could it be damaged (brand new) or is it more likely to be a PCI-BIOS issue? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE205P leds flash red on startup
--- On Thu, 12/15/11, A J Stiles asterisk_l...@earthshod.co.uk wrote: I have a new Digium TE205P 2-span E1 card I just installed on a server. As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? No, it's normal behaviour until the card's firmware has been loaded. Which can't happen until the kernel is booted; and probably will not happen until the Zaptel or DAHDI startup script runs. Well, strange enough, the server used to have a single-span PRI card, booted with kernel 2.6.23 and autoloaded the appropriate zaptel 1.4.12.1 module (wcte12xp). Now I replaced the single-span card with the dual-span TE205 and rebooted. The kernel does not autoload the new zaptel module which should be wct4xxp. So I try to load it manually (modprobe -a wct4xxp) and lsmod lists it but there's nothing in /proc/zaptel/. I suppose the 1205 identifier is correct for the TE205 card, as seen after issuing lspci: 05:01.0 Communication controller: Digium, Inc. Unknown device 1205 (rev 02) Subsystem: Unknown device 0005: Flags: bus master, medium devsel, latency 64, IRQ 5 Memory at feaefc00 (32-bit, non-prefetchable) [size=128] I left my zaptel.conf and zapata.conf files untouched as, theoretically, they should work just fine, at least for the first PRI port on the card (everything else is identical). So zaptel.conf has something like this: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 and zapata.conf: switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-15 channel = 17-31 However, if I run ztcfg I get this message: ZT_SPANCONFIG failed on span 1: No such device or address (6) The fact that there's nothing in /proc/zaptel/ makes me think that the zaptel kernel module isn't working. Is the 1205 card compatible with zaptel 1.4.12.1? (I can't migrate to DAHDI on this system - at least not yet) Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRI configuration
Interesting: If you cannot obtain T1 specific cable, then use two runs of CAT 5. Use one CAT5 cable for the Transmit (Tx) signal and one CAT5 cable for the Receive (Rx) signal. It is necessary for the Tx and Rx signals to be in separate sheaths to prevent cross talk interference So pins 1 and 2 on one cable and pins 4 and 5 on another. --- On Thu, 12/8/11, Lyle Giese l...@lcrcomputer.net wrote: Try this instead: http://www.ahk.com/t1_cable.html That cisco link does not specify the cable itself, but only the pin outs. True T1 cable has a foil shield around each pair, also called ABAM cable in the telco world. Ethernet cable is twisted pair without any shielding between pairs. And one shield around all the pairs is not the same as ABAM. Lyle Giese LCR Computer Services, Inc. On 12/08/11 10:53, Carlos Alvarez wrote: A T1 cable according to this spec: http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml Crossing the 1/2 to 4/5 if needed. On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr mailto:oza_4...@yahoo.fr wrote: 2011/12/8, Carlos Alvarez car...@televolve.com mailto:car...@televolve.com: I am not Kevin, but I'll tell you that I will not EVER use an Ethernet cable for T1 again. Kevin and I have discussed this at length, and the should work plays out poorly in the real world, or at least mine. I've had it be fine, and had major problems. I can't even find a pattern to it, like length of cable. In a colo cabinet that was direct-connected to a carrier, it worked great for years and then one day...no T1. Just gone. Go down there and put in a real T1 cable, came right up, still up years later. I usually make my own, which type of cable are you then using ? since they are so expensive to buy. I just connect the four needed pins, pretty easy to do if you're not trying to stuff all eight wires into the connector. On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield t...@softins.co.uk mailto:t...@softins.co.uk wrote: In article 4ee0b0e2.3050...@digium.com mailto:4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk mailto:t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org mailto:t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 tel:602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] ss7 installation and configuration
Hi, I'm unable to configure SS7 (surely my bad because it's my first try). I get this error: ERROR[15475] chan_dahdi.c: Unknown signalling method 'ss7' My system has: asterisk 1.4.31 zaptel 1.4.12.1 libpri 1.4.11.5 libss7 1.0.1 (installed from source) I can't upgrade this server to Dahdi and latest asterisk version... In any case, according to the libss7 README, it should work with my software versions. How can I make sure Asterisk is loading the SS7 library? According to libss7, I should place signalling=ss7 in /etc/asterisk/zapata.conf. Is that right? Do I need to recompile zaptel AFTER I install libss7? Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN PRI configuration
Hi, A telco has recently installed a new line in our building and I need to connect it to my Asterisk server with a Digium PRI card. It's not the first time I set up and configure a PRI link but I'm failing to make this one work. The only information I got from the telco is: Line Coding [HDB3] Framing [CRC4] Encapsultation [hdlc Isdn switch-type primary-[net5] Is crc4 actually a framing parameter as stated by the telco, or is it just an optional line coding parameter? I searched the web and not knowing exactly which parameters to use, I tried the following zaptel/dahdi config: # TE120P (PRI): span=1,1,0,ccs,hdb3,crc4 # as E1 bchan=1-15 dchan=16 bchan=17-31 switchtype = euroisdn signalling = pri_cpe However, the link doesn't work and I get this: *CLI show status: Description Alarms IRQbpviol CRC4 Wildcard TE120P Card 0 RED1 0 0 # cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE120P Card 0 (MASTER) HDB3/CCS/CRC4 RED IRQ misses: 1 1 WCT1/0/1 Clear (In use) RED 2 WCT1/0/2 Clear (In use) RED 3 WCT1/0/3 Clear (In use) RED 4 WCT1/0/4 Clear (In use) RED 5 WCT1/0/5 Clear (In use) RED 6 WCT1/0/6 Clear (In use) RED 7 WCT1/0/7 Clear (In use) RED 8 WCT1/0/8 Clear (In use) RED 9 WCT1/0/9 Clear (In use) RED 10 WCT1/0/10 Clear (In use) RED 11 WCT1/0/11 Clear (In use) RED 12 WCT1/0/12 Clear (In use) RED 13 WCT1/0/13 Clear (In use) RED 14 WCT1/0/14 Clear (In use) RED 15 WCT1/0/15 Clear (In use) RED 16 WCT1/0/16 HDLCFCS (In use) RED 17 WCT1/0/17 Clear (In use) RED 18 WCT1/0/18 Clear (In use) RED 19 WCT1/0/19 Clear (In use) RED 20 WCT1/0/20 Clear (In use) RED 21 WCT1/0/21 Clear (In use) RED 22 WCT1/0/22 Clear (In use) RED 23 WCT1/0/23 Clear (In use) RED 24 WCT1/0/24 Clear (In use) RED 25 WCT1/0/25 Clear (In use) RED 26 WCT1/0/26 Clear (In use) RED 27 WCT1/0/27 Clear (In use) RED 28 WCT1/0/28 Clear (In use) RED 29 WCT1/0/29 Clear (In use) RED 30 WCT1/0/30 Clear (In use) RED 31 WCT1/0/31 Clear (In use) RED Placing a call through the Zap/Dahdi trunk in Asterisk doesn't work and I get the following message in the log: chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! logger.c: -- Attempting call on Zap/g1/999xx for 999xx@custom-TESTCALL:1 (Retry 1) channel.c: Unable to request channel Zap/g1/999xx pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy) chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! Am I missing some information here? I'm *supposing* it should be E1 (and that I can use 16 as dchan), euroisdn (not national), but my telco states hdlc Isdn switch-type primary-[net5] and I don't know how to translate it to zaptel/dahdi... Also, my telco hasn't mentioned anything about ccs but I tried it anyway because I wouldn't know what else to use. I also tried signalling = pri_net but still got the same RED alerts. Any suggestions? Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRI configuration
--- On Wed, 12/7/11, Steve Edwards asterisk@sedwards.com wrote: A telco has recently installed a new line in our building and I need to connect it to my Asterisk server with a Digium PRI card. It's not the first time I set up and configure a PRI link but I'm failing to make this one work. chan_dahdi.c: No D-channels available! Using Primary channel 16 as D-channel anyway! We usually get D channels on the first channel of the first T1 in an NFAS group and the last channel of the last t1. However, telcos don't always get the order right. I've spent hours trying configurations and varying the D channel. Sometimes it's just that they number things in a different order than we were expecting. Sometimes, it almost appears that they use a dartboard :) As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you suggested and set 1 as the D channel and 2-31 as B channels. In the asterisk log I got these messages: chan_dahdi.c: Channel 16 is reserved for D-channel. chan_dahdi.c: Unable to register channel '2-31' So doesn't this actually tell me that I should keep using 16 as the D channel? (so chan_dahdi actually knows about it on its own, I guess) It's funny though that chan_dahdi tells me I have to use channel 16 as D channel whenever I try to use another one, but when I do use 16, it says that there are no D channels available. Confusing. Thanks anyway for the reply. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRI configuration
--- On Wed, 12/7/11, Kevin P. Fleming kpflem...@digium.com wrote: Vieri: You aren't even far enough along to worry about D-channel assignments or anything like that. Your span is in RED alarm; that means it can't see the far end at all. Until you get that cured (layer 1 - physical layer) nothing above it is going to work. Since they mentioned HDB3 and CRC4, you most definitely have an E1 span, and you will need to specify 'CCS' as well because you are using ISDN signaling. If the line coding/framing settings are wrong that *could* result in a RED alarm, but doesn't always. So, you need to start by getting the span to come out of RED alarm (to go 'green'). This could be a cabling problem, a hardware problem, or it could something as simple as the fact that the telco hasn't actually 'turned up' the span yet, because they don't usually do that until you have your equipment plugged in and you call them to tell them that you are ready for the span to be turned up. They should have turned it up, or at least that's what one of the tech guys told me. But I guess I'll have to check with them again. The cable should be ok (standard ethernet cable) but I didn't actually install it myself (I'm in a remote location) so I'll have to check that too. Big thanks for the explanation! Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk: BYE is received late
Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call is correctly established and conversation is OK. If the local softphone user hangs up first, the remote end is also disconnected immediately. However, if the remote party hangs up first, the local caller is not immediately disconnected. That, of course, is undesirable. I'd like to understand why the call isn't automatically hung up and fix it. I'm supposing that Jitsi isn't receiving a BYE as expected in a correct SIP transaction (or BYE is arriving very late). I don't know why though. Here's my network setup: Softphone asterisk extension 4053 at 10.215.144.48 Asterisk eth0: 10.215.147.111 but softphone registers to the alias/floating IP for failover setup 10.215.147.115 Asterisk eth1: 192.168.103.111 Asterisk default gateway: 192.168.103.1 - Asterisk accesses Internet via eth1 (192.168.103.1 is a DSL modem/router) I did a tcpdump on the asterisk server while calling from the local softphone as so: tcpdump -s0 -X -n -w asterisk.cap -i eth0 host 10.215.144.48 It's here: http://213.96.91.201/temp/jitsi_via_asterisk.cap.gz Here's the full session (softphone waits 2 minutes until it finally hangs up): http://213.96.91.201/temp/jitsi_via_asterisk_full_session.cap.gz Asterisk seems to send BYE to the softphone after 120 seconds since the remote party actually hung up... A packet dump on eth1 during the call also shows the BYE message coming in from the SIP provider: http://213.96.91.201/temp/asterisk_eth1.txt I'm almost certain the remote SIP provider sends BYE in time because earlier today I tested by connecting the softphone directly to the SIP provider and going out the same DSL line (thus removing Asterisk from the equation). ie. I placed a laptop with Jitsi in the same subnet 192.168.103.0 and used the default gateway 192.168.103.1 (just like Asterisk). All went well. I also setup my Jitsi laptop within the 10.215.0.0 subnet (just like my Asterisk client setup) but connected directly to the SIP provider (without going through Asterisk). In this case the call ended as expected (OK). So I guess that something's wrong with my Asterisk configuration. Both my softphone and network configuration *should* be OK. However, it may have something to do with my Asterisk eth0/eth1 setup but I don't see what. Any ideas/suggestions? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk: BYE is received late
For the record, it seems to be a SIP-ALG issue. It's fixed now. Vieri --- On Wed, 6/8/11, Vieri rentor...@yahoo.com wrote: Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call is correctly established and conversation is OK. If the local softphone user hangs up first, the remote end is also disconnected immediately. However, if the remote party hangs up first, the local caller is not immediately disconnected. That, of course, is undesirable. I'd like to understand why the call isn't automatically hung up and fix it. I'm supposing that Jitsi isn't receiving a BYE as expected in a correct SIP transaction (or BYE is arriving very late). I don't know why though. Here's my network setup: Softphone asterisk extension 4053 at 10.215.144.48 Asterisk eth0: 10.215.147.111 but softphone registers to the alias/floating IP for failover setup 10.215.147.115 Asterisk eth1: 192.168.103.111 Asterisk default gateway: 192.168.103.1 - Asterisk accesses Internet via eth1 (192.168.103.1 is a DSL modem/router) I did a tcpdump on the asterisk server while calling from the local softphone as so: tcpdump -s0 -X -n -w asterisk.cap -i eth0 host 10.215.144.48 It's here: http://213.96.91.201/temp/jitsi_via_asterisk.cap.gz Here's the full session (softphone waits 2 minutes until it finally hangs up): http://213.96.91.201/temp/jitsi_via_asterisk_full_session.cap.gz Asterisk seems to send BYE to the softphone after 120 seconds since the remote party actually hung up... A packet dump on eth1 during the call also shows the BYE message coming in from the SIP provider: http://213.96.91.201/temp/asterisk_eth1.txt I'm almost certain the remote SIP provider sends BYE in time because earlier today I tested by connecting the softphone directly to the SIP provider and going out the same DSL line (thus removing Asterisk from the equation). ie. I placed a laptop with Jitsi in the same subnet 192.168.103.0 and used the default gateway 192.168.103.1 (just like Asterisk). All went well. I also setup my Jitsi laptop within the 10.215.0.0 subnet (just like my Asterisk client setup) but connected directly to the SIP provider (without going through Asterisk). In this case the call ended as expected (OK). So I guess that something's wrong with my Asterisk configuration. Both my softphone and network configuration *should* be OK. However, it may have something to do with my Asterisk eth0/eth1 setup but I don't see what. Any ideas/suggestions? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pipe audio stream to external application
Hi, I'd like to know if there's an easy way of doing the following: SIP phone dials a custom feature code in Asterisk, call gets answered within a custom context (Answer()), anything that the caller says should be redirected/piped to an external application. Something like monitor except audio should be sent live. More like app_ices (or app_ezstream if that existed) but for a generic app. Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail-over server
--- On Thu, 2/10/11, Jonathan Thurman jonat...@thurmantech.com wrote: Have you looked at the 'defaultip' sip configuration option? Or setting host=IP for those devices? I've read that defaultip can only be used on type=peer and when host=dynamic. I use type=friend. host=IP seems to be OK for me. I actually tried this option some time ago but had trouble with something I can't recall right now so reverted to dynamic. I guess I'll have to give it another shot. I'll try that before migrating to realtime... Thanks Jonathan! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail-over server
--- On Tue, 2/8/11, Jonathan Thurman jonat...@thurmantech.com wrote: It depends on your configuration. If you use Asterisk Realtime to store SIP registrations, then the database will contain information on how to contact the device (fullcontact, ipaddr, and port fields). Then on a failover, Asterisk will do a lookup for the peer in the database, find the needed information and dial the device. I don't use realtime and haven't tried it yet. I don't know much about the SIP protocol but can't the server send a notification of some sort to peers so as to quicken re-registration? I'm thinking of something similar to sip notify. Since all of the SIP devices in my LAN have static IP addresses, I can keep track of everyone on my own. For instance, could I do fake SIP registrations from localhost (the * server) and specify a LAN IP address? I would write a custom script that would execute whenever an Asterisk server takes over. As said earlier, this server would not have any SIP extensions registered at first and they would be registering slowly within 60 seconds or more. However, since I KNOW FOR SURE that some SIP devices are always online and have static IP addresses, can't I fool Asterisk by somehow registering via locahost but spoofing the source IP address? Maybe setting the source port to what it was exactly can be tougher but I *could* try to keep track of it. This way, whenever the Asterisk server that took over tries to bridge a call, it will try to connect to the fakely-registered IP address. I'm not using realtime for 2 reasons: 1- I'm using the FreePBX framework and there's no realtime backend unfortunately. Moving to Realtime and losing all the FreePBX goodies is time-consuming. Does anyone know how to use FreePBX + Realtime? 2- I don't have enough hardware resources to setup a server for the realtime DB that both Asterisk servers would connect to. Also, I wouldn't feel comfortable having just one DB server. For easier maintenance I would use a clustered database for realtime. However, I'm using Mysql 5.0 ndbcluster tables for other non-voip purposes and my experience hasn't been so great. I once had a power outage and all ndb table data was lost. Also, 5.0 ndb crashes in several occasions. As far as I can tell, it isn't reliable. I haven't tried 5.1 though. I have no experience with clustered postgresql. In the above scenario, I can kill Asterisk, start it again, and place a call from two devices that have not registered again. I'd like to do that without Realtime (or with Realtime+FreePBX) or with any other means that doesn't require more than 2 servers (2 asterisk boxes)? Feedback appreciated. Thanks Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP registration
Hi, Are sip.conf's defaultexpiry and maxexpiry global? Or can they be used on a per-extension basis? I'd like to force some extensions to re-register more frequently than others (server-side). Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fail-over server
Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually on-line so calls can be routed to them. How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exceptionally long queue length queuing . . . .
I have the same problem, once in a while. Curiously though, it occurs on a dedicated 100Mbps switched local network. I'm running 1.4.31 * servers. Vieri --- On Sat, 10/30/10, Brian Capouch bri...@palaver.net wrote: I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of the bug tracker tickets that it is instigated by a small network leg looks to be on point to me, because while it happens way often, it doesn't always happen. My ITSPs have all dropped IAX, and if they're experiencing this problem I can see why. Once the first of these messages has occurred, it's goodbye audio for the rest of the call. If anyone has a perspective on this longstanding problem, I'd sure be glad to hear it. Thanks. b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opensource Speech recognition for Asterisk
Hi, Sorry to drop in on this thread but I'm relatively new to Sphinx and speech recognition. I'd like to know if anyone has successfully setup speech recognition in Asterisk for Spanish users. Sphinx doesn't seem to have Spanish acoustic and language models and I don't think I'll ever have the time or know-how to make my own. My requirements are similar to the OP's: basic yes, no, get an 8 digit number, etc. Actually, yes (si) and no work well with the English models. However, accuracy is not that great when it comes to recognizing digits zero to nine in Spanish. Thanks for any suggestions, Vieri --- On Tue, 8/24/10, Bob Kleiner bob.klei...@gmail.com wrote: From: Bob Kleiner bob.klei...@gmail.com Subject: Re: [asterisk-users] Opensource Speech recognition for Asterisk To: asterisk-users@lists.digium.com Date: Tuesday, August 24, 2010, 7:30 AM Thanks guys. A lot of info here :-) I am wondering if anyone followed this and it was working for them: http://scribblej.com/svn/ ??? Hello Bruce We successfully deployed it and now saving thousands on commercial ASR ports. It seems users are rather happy with it. The recognition seems pretty accurate. Of course it has it's own limitations but so any other technology. It will not hurt if some of your users will benefit from ASR. I am not looking for anything fancy. The basic yes, no, dialing a number, asking for agent, etc...out of which probably the hardest is a 10 digit number to be asked to be dialed. Yes, that should work. It also supports JSGF grammars, so you should be able to recognize digit strings easily. And if you want something serious, there are at least two open source products providing ASR over standard MRCP protocol. They also use CMUSphinx, so provide the same accuracy Zanzibar http://www.spokentech.org/writing-speechlets.html Cairo http://www.speechforge.org/ Though Cairo is a bit dated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk, Skype For SIP
--- On Mon, 7/19/10, Kevin P. Fleming kpflem...@digium.com wrote: Usage of the standard Skype client is not free; it involves acting as part of the peer-to-peer Skype network The Skype business solutions (including Skype For Asterisk) don't participate in the peer-to-peer network Any solution that uses a regular Skype client will be limited to one call at a time; Thanks for the explanation! It's crystal-clear now. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype for Asterisk, Skype For SIP
Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype user. I do not need to call landlines via Skype. 2) allow Internet Skype users to call my Asterisk PBX Skype user and route the call to a specific Asterisk SIP extension. At first, I thought it would be simple and free. However, correct me if I'm wrong but the Skype user I can use within the Asterisk PBX cannot be the standard type (used by eg. desktop Skype applications) but needs to be created by the Skype User Manager for Business Solutions. I believe this has a price although Skype For SIP Open Beta seems to be free until Q4 2010. Has anyone found a way to make pure Internet user-to-user Skype/SIP calls via Asterisk (no PSTN involved) for free? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk, Skype For SIP
--- On Sun, 7/18/10, Alejandro Imass a...@p2ee.org wrote: Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype user. I do not need to call landlines via Skype. I think this is _explicitly_ not supported in the Skype for SIP docs. 2) allow Internet Skype users to call my Asterisk PBX Skype user and route the call to a specific Asterisk SIP extension. Here is how it goes from my experience with Skype: each SIP channel will cost you about $5 a month, regardless if you have a landline number with them or not. Your account will be assigned a special Skype number 99x . With that number a Skype user can call you and it will be free. You _cannot_ call Skype users from your PBX, as I stated above, this is an explicit no-no in the docs. If you want to make calls from your PBX to landlines you have to buy Skype credit just like you would with a regular skype client. If you want land-lines to call your PBX you need to purchase a skype number which about $60 a year. At first, I thought it would be simple and free. However, correct me if I'm wrong but the Skype user I can use within the Asterisk PBX cannot be the standard type (used by eg. desktop Skype applications) but needs to be created by the Skype User Manager for Business Solutions. I believe this has a price although Skype For SIP Open Beta seems to be free until Q4 2010. I think you can associate existing skype users to your Business Solutions manager but I still don't understand exactly how or why this is useful, and I don't think it has to do with you being able to call any of them from your PBX. Then again I haven't paid much attention to that and perhaps you have more insight into this. Has anyone found a way to make pure Internet user-to-user Skype/SIP calls via Asterisk (no PSTN involved) for free? As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. Thanks Alejandro, I still don't see why one should pay for a channel when using a PBX but not when using a client such as Skype. OK, I know that the Skype network is proprietary and I have to accept whatever they say. However, if a standard user can call and receive for free then there should be a way to do it from a PBX such as Asterisk. In fact, I came across this project: http://www.mhspot.com/sts/siptosis.html It seems to be a bit of a hack in that it integrates a SIP PBX with a standard Skype client (which doesn't necessarily have to be on the same machine or same OS...). In short, one can use a standard Skype account and not pay a cent for user-to-user calls. Can chan_skype do that? (it doesn't seem to) Has anyone tried SipToSis? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_iax2: I should never be called!
Hi, Recently, one of my Asterisk servers stopped connecting calls and required a reboot to fix it (did not try to restart or reload). The log showed loads of this message: NOTICE[302] chan_iax2.c: I should never be called! This highly repeated message seems to be preceded by something like: WARNING[10767] channel.c: Exceptionally long voice queue length queuing to IAX2/coinbound-15879 When this happens it also seems that SIP peers on a gigabit LAN start going on/offline frequently. So that seems to explain why calls start to fail. There is absolutely nothing wrong with the network (and switches). I don't know if it can be a NIC problem on the server but how can I tell? [Jul 9 08:10:49] NOTICE[10756] chan_sip.c: Peer '7054' is now Lagged. (2819ms / 2000ms) [Jul 9 08:10:50] NOTICE[10756] chan_sip.c: Peer '7054' is now Reachable. (860ms / 2000ms) [Jul 9 08:10:51] NOTICE[10756] chan_sip.c: Peer '7054' is now Lagged. (2003ms / 2000ms) [Jul 9 08:10:52] NOTICE[10756] chan_sip.c: Peer '7054' is now Reachable. (876ms / 2000ms) [Jul 9 08:10:54] NOTICE[10756] chan_sip.c: Peer '7054' is now Lagged. (2929ms / 2000ms) [Jul 9 08:10:56] NOTICE[10756] chan_sip.c: Peer '7054' is now Reachable. (963ms / 2000ms) [Jul 9 08:11:03] NOTICE[10756] chan_sip.c: Peer '7054' is now UNREACHABLE! Last qualify: 3096 Rebooting the server solved everything... for now... Any ideas? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR
How can I match any_num_of_digits#any_num_of_digits in an IVR?
I want users to be able to type, eg., 123#4567
I tried the following but it hangs up immediately. If I uncomment WaitExten
then it hangs up right when the user dials #.
As a side question, can I play a background message while using the Read()
command?
[FILTER-validate]
exten = h,1,Hangup()
exten = hang,1,Hangup()
exten = s,1,Set(CANCALL=1)
exten = s,n,Set(LOOPCOUNT=0)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=5)
exten = s,n(repeatme),Background(TEST/FILTER_VALIDATE_1)
;exten = s,n,WaitExten(5,m(default))
exten = _X.#XXX.,1,Playback(one-moment-please)
exten = _X.#XXX.,n,AGI(filter-validate.agi|${EXTEN})
exten = _X.#XXX.,n,GotoIf($[${CANCALL} =
1]?outbound,${CANCALL_EXTEN},filterok)
exten = _X.#XXX.,n,Playback(TEST/FILTER_VALIDATE_3)
exten = _X.#XXX.,n,Hangup()
exten = t,1,Hangup()
exten = i,1,Playback(invalid)
exten = i,n,Goto(loop,1)
exten = loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten = loop,n,GotoIf($[${LOOPCOUNT} 2]?hang,1)
exten = loop,n,Goto(FILTER-validate,s,repeatme)
Thanks,
Vieri
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[asterisk-users] AGI library for C/C++
I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6 compatible). I've taken a look at CAGI and QUIVR but their latest code releases date back to 2006. I've also seen a more recent project (wildpbx) dated 2009: http://github.com/comradeb14ck/wildpbx/tree/master/libraries/agi/c/ Any suggestions/recommendations for a C AGI library? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] get Asterisk version from within dialplan
Simple enough: How can I get Asterisk version from within my dialplan? (preferably without calling an AGI script that parses asterisk -rx show version) Is it available as a global variable? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
It happens even with just a few calls (way less than 30). I'm trying to see if Asus has something to say about this. In the meantime I'm using trunk=no and it's working fine. Thanks Vieri --- On Fri, 5/14/10, Zoa zoach...@securax.org wrote: I think that the clock resets would cause no audio or garbled audio every 20 minutes, not constant interference. Could you tell us how many simultaneous calls were in the trunk and what the size is of 1 voice packet ? Can you try putting maximum 30 calls per trunk (use multiple trunks if needed) and see if the problem goes away. Greetings, zOa Vieri wrote: --- On Thu, 5/13/10, Zoa zoach...@securax.org wrote: Can you try trunk = no ? Lifesaver... trunk=no made the interference go away. I have clean audio now. Quote: IAX Trunking needs support of a hardware timer. I'm supposing my system is using the DAHDI-driven Digium cards on my motherboard. I don't know how hardware timers work and if Digium hardware rely on the motherboard (my system clock is going too fast and my ntpd is constantly adjusting the clock by -2.6 seconds every 20 minutes). In any case, since I'm on a dedicated LAN I guess I can safely set trunk=no. Thanks! Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
--- On Fri, 5/14/10, Steve Edwards asterisk@sedwards.com wrote: I'm supposing my system is using the DAHDI-driven Digium cards on my motherboard. I don't know how hardware timers work and if Digium hardware rely on the motherboard (my system clock is going too fast and my ntpd is constantly adjusting the clock by -2.6 seconds every 20 minutes). In any case, since I'm on a dedicated LAN I guess I can safely set trunk=no. Maybe it's just me, but I'd be thinking if the mobo manufacturer did such a crappy job on the clock, what else is wrong. I'd be looking for a better mobo. The manufacturer is ASUS. The mobo is M4A77TD PRO, latest BIOS update. Supposedly manufacturers such as HP and Dell should be better but people usually have a good opinion on Asus. Beats me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
--- On Thu, 5/13/10, Zoa zoach...@securax.org wrote: Can you try trunk = no ? Lifesaver... trunk=no made the interference go away. I have clean audio now. Quote: IAX Trunking needs support of a hardware timer. I'm supposing my system is using the DAHDI-driven Digium cards on my motherboard. I don't know how hardware timers work and if Digium hardware rely on the motherboard (my system clock is going too fast and my ntpd is constantly adjusting the clock by -2.6 seconds every 20 minutes). In any case, since I'm on a dedicated LAN I guess I can safely set trunk=no. Thanks! Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]
--- On Fri, 5/14/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: You were probably caught be the fact that you are using extension numbers also as SIP user names for your phones (here: 3666). This is not a good thing to do, better use an alphanumeric username or the phone's MAC address etc. Is there more info on this? I mean, why is it bad, apart from the security implication. As for your IAX sound quality issue: I have seen that before as well, and switched to SIP (as others did). My guess is that it will probably go away if you use Asterisk 1.4 on both sides, though. It went away even with 1.2 but I needed to set trunk=no. Probably a jitter buffer issue on my system(s). SIP DEBUG on the receiving Asterisk gives you a hint which peer was found if matching is done on the IP address, the text is somethint like Found peer ... or Found no matching peer or user for w.x.y.z Tnanks for the info Philipp. I'll try to further debug my SIP messages. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers [SOLVED]
Issue solved. Looks like all I was missing was one parameter: fromuser= Thanks for your time! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a third noise overlapping with a scratchy sound as if it were some kind of interference). So lately I setup calls to go through the SIP trunk and audio quality is OK (no third overlapping noise). This is happening between Asterisk 1.4.31 and a 1.2.40. I'm wondering if there's something I can tweak in IAX2 to eliminate this artifact. Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's enabled by default)? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
--- On Thu, 5/13/10, Gareth Blades list-aster...@skycomuk.com wrote: Show the details on the active channels when using both methods and check what codecs are being used. The audio codecs are different: Type: SIP State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) WriteTranscode: Yes ReadTranscode: Yes Type: IAX2 State: Up (6) Rings: 0 NativeFormats: 0x8 (alaw) WriteFormat: 0x8 (alaw) ReadFormat: 0x8 (alaw) WriteTranscode: No ReadTranscode: No By the way, I have this in iax.conf: [interboxIAX2] deny=all allow=ulaw allow=gsm type=friend host=192.168.250.111 secret=mysecret auth=plaintext requirecalltoken=no qualify=yes context=mycontext trunk=yes username=interbox Shouldn't the channel details report ulaw instead of alaw? Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is the same (I still experience bad audio quality). Maybe I should try slin but how do I force it? Vieri wrote: Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a third noise overlapping with a scratchy sound as if it were some kind of interference). So lately I setup calls to go through the SIP trunk and audio quality is OK (no third overlapping noise). This is happening between Asterisk 1.4.31 and a 1.2.40. I'm wondering if there's something I can tweak in IAX2 to eliminate this artifact. Could the IAX2 jitter buffer between 1.2 and 1.4 be an issue (I believe it's enabled by default)? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP trunk between two Asterisk servers
Hi, I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls). With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious. I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/. so Asterisk server 1 (192.168.250.111) sip.conf contains: [interboxsip] type=peer host=192.168.250.112 context=mycontext Asterisk server 2 (192.168.250.112) sip.conf contains: [interboxsip] type=peer host=192.168.250.111 context=mycontext I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666 in server 1 (192.168.250.111) via the interboxsip SIP trunk. The call fails and according to the SIP messages it seems to be an authentication problem. What am I missing? SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call): -- Executing [3...@from-internal:2] Dial(SIP/4053-6dea, SIP/interboxsip/3666|300|rt) in new stack Audio is at 192.168.250.112 port 15850 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.250.111:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport From: device sip:4...@192.168.250.112;tag=as4d17a185 To: sip:3...@192.168.250.111 Contact: sip:4...@192.168.250.112 Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:13:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 15850 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called interboxsip/3666 --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060 From: device sip:4...@192.168.250.112;tag=as4d17a185 To: sip:3...@192.168.250.111;tag=as00842b82 Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2545a5dd Content-Length: 0 - --- (10 headers 0 lines) --- Transmitting (no NAT) to 192.168.250.111:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport From: device sip:4...@192.168.250.112;tag=as4d17a185 To: sip:3...@192.168.250.111;tag=as00842b82 Contact: sip:4...@192.168.250.112 Call-ID: 3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/interboxsip-6deb is circuit-busy SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111 Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Contact:
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Either a) set a secret and use that on both sides, or b) look at allowguest= and the default context and maybe the domain= settings, or c) use insecure=invite Thanks Philipp. I'm trying option c) which is the simplest. used insecure=invite but failed with the same SIP messages. Tried also insecure=yes but the same messages show up: SIP/2.0 407 Proxy Authentication Required I had already tried a) before but did not record the SIP messages (it also failed). I haven't tried c) yet... So I'll do a) again and log the messages and then try c). Do you actually have a working SIP trunk within your LAN? If so, could you please share your settings? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote:
I have forget to write for outcall in
extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten =
_X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten =
_X.,2,Dial(SIP/interboxserver1/${EXTEN})
exten = _X.,3,Hangup
:)
Vardan
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver1]
exten = _X.,1,Noop(Call from server1)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Try so, I think it must work.
And also, look and delete any another records in both
servers in
sip.conf about this servers settings.
Vardan
Vieri wrote:
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk
servers on the same LAN (no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP.
I must be missing something obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.112
context=mycontext
Asterisk server 2 (192.168.250.112) sip.conf
contains:
[interboxsip]
type=peer
host=192.168.250.111
context=mycontext
I dialed from a SIP extension (4053) in server 2
(192.168.250.112) to 3666 in server 1 (192.168.250.111) via
the interboxsip SIP trunk.
The call fails and according to the SIP messages
it seems to be an authentication problem.
What am I missing?
SIP messages on 192.168.250.112 (Asterisk server 2
- transmitting call):
-- Executing
[3...@from-internal:2] Dial(SIP/4053-6dea,
SIP/interboxsip/3666|300|rt) in new stack
Audio is at 192.168.250.112 port 15850
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to
192.168.250.111:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:13:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 15850 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called
interboxsip/3666
--- SIP read from 192.168.250.111:5060
---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5,
realm=asterisk, nonce=2545a5dd
Content-Length: 0
-
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.250.111:5060:
ACK sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
From:
devicesip:4...@192.168.250.112;tag=as4d17a185
To:sip:3...@192.168.250.111;tag=as00842b82
Contact:sip:4...@192.168.250.112
Call-ID:
3770a8004ce882dd3c89c1d91b5aa...@192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
--
SIP/interboxsip-6deb is circuit-busy
SIP messages on 192.168.250.111 (Asterisk server 1
- receiving end):
-- SIP read from 192.168.250.112:5060:
INVITE sip:3...@192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP
192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From:
devicesip:4...@192.168.250.112;tag=as18a568d6
To:sip:3...@192.168.250.111
Contact:sip:4...@192.168.250.112
Call-ID:
328617546726e5d430538e8061771...@192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111 Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: device sip:4...@192.168.250.112;tag=as18a568d6 To: sip:3...@192.168.250.111;tag=as57a19dac Contact: sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: SIP/2.0 407 Proxy Authentication Required Then you have another entry in sip.conf that uses the same IP address. Delete that, or change the port on one of them, and adjust insecure= accordingly. asterisk1 # grep 192.168.250 sip*.conf sip.conf:host=192.168.250.112 asterisk2 # grep 192.168.250 sip*.conf sip.conf:host=192.168.250.111 So I only have 1 entry in each server's sip.conf and this entry is in interboxsip (my sample SIP trunk name). Puzzling... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (90 ms) interboxsip192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7D N A 13404OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6D N A 25967OK (10 ms) 7118/7118 192.168.250.10 D N A 14508OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8D N A 12342OK (10 ms) 7112/7112 192.168.250.31 D N A 19829OK (10 ms) 7111/7111 192.168.250.32 D N A 35259OK (80 ms) 7109/7109 (Unspecified)D N A 0UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username HostDyn Nat ACL Port Status sipprovider/01 w.x.y.zN 5060 OK (79 ms) interboxsip192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: And sip show registry sip show registry doesn't list anything regarding my interboxsip test trunk because I'm trying to setup a straightforward link such as this one described here (without user/password): http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ The only sip show registry entry I have is the one for my external Internet SIP trunk, which is ok. Thanks for your time. Vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: What are your allowguest= and domain= settings in the global section of sip.conf? And which version of Asterisk exactly are you using? I have no such settings defined yet. Still haven't tried to set them... Not sure what to put in domain. Anyway: # /etc/asterisk/sip.conf [general] vmexten=*97 disallow=all allow=ulaw allow=alaw context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 rtptimeout=120 rtpholdtimeout=300 pedantic=no urlencode=yes register=01:...@internet_sip_provider.com/01010101010101 regcontext=dundi-extens Server 2: Asterisk 1.4.31 Server 1: same sip.conf settings except Asterisk 1.2.40 Notice the urlencode setting which is a patch taken from: https://issues.asterisk.org/view.php?id=14652 This may be the culprit but I'm not quite sure about it. Also, I *need* this patch unless the address incomplete issue gets solved. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP trunk between two Asterisk servers
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote: Please change the peers name in any server. for example: server1: interboxsip1 server2: interboxsip2 If I understand correctly, the peer names can be identical on both servers. What counts is the host entry, I guess. But then again, my SIP trunk isn't working so I'll try out your suggestion tomorrow. Thanks, Vieri Vardan Vieri wrote: --- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote: please show sip show users and sip show peers SERVER 2: sip show users (trimmed to just my sip test trunk): Username Secret Accountcode Def.Context ACL NAT interboxsip mycontext No RFC3581 sip show peers (also trimmed): Name/username Host Dyn Nat ACL Port Status sipprovider/01 w.x.y.z N 5060 OK (90 ms) interboxsip 192.168.250.111 5060 Unmonitored 7503/7503 10.215.146.190 D N A 5060 OK (20 ms) 7502/7502 10.215.146.203 D N A 5060 OK (20 ms) 7172/7172 192.168.250.7 D N A 13404 OK (40 ms) 7166/7166 10.215.146.200 D N A 5060 OK (20 ms) 7165/7165 10.215.248.12 D N A 5060 OK (1 ms) 7160/7160 10.215.146.182 D N A 5060 OK (20 ms) 7137/7137 192.168.250.6 D N A 25967 OK (10 ms) 7118/7118 192.168.250.10 D N A 14508 OK (1 ms) 7117/7117 10.215.146.185 D N A 5060 OK (20 ms) 7114/7114 192.168.250.8 D N A 12342 OK (10 ms) 7112/7112 192.168.250.31 D N A 19829 OK (10 ms) 7111/7111 192.168.250.32 D N A 35259 OK (80 ms) 7109/7109 (Unspecified) D N A 0 UNKNOWN 7097/7097 10.215.146.164 D N A 5060 OK (20 ms) SERVER 1: sip show users is identical. sip show peers (trimmed): Name/username Host Dyn Nat ACL Port Status sipprovider/01 w.x.y.z N 5060 OK (79 ms) interboxsip 192.168.250.112 5060 Unmonitored vardan Vieri wrote: --- On Wed, 5/12/10, Philipp von Klitzingklitz...@pool.informatik.rwth-aachen.de wrote: --- SIP read from 192.168.250.111:5060 --- SIP/2.0 407 Proxy Authentication Required You need to run the SIP debug on 192.168.250.111 to learn more about WHY the 407 is issued. Have a close look and you are likely to understand it right away. Also: Do not forget the reload after applying changes to sip.conf. I always do a sip reload after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): -- SIP read from 192.168.250.112:5060: INVITE sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111 Contact:sip:4...@192.168.250.112 Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - 328617546726e5d430538e8061771...@192.168.250.112 Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: devicesip:4...@192.168.250.112;tag=as18a568d6 To:sip:3...@192.168.250.111;tag=as57a19dac Call-ID: 328617546726e5d430538e8061771...@192.168.250.112 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1327c5b6 Content-Length: 0 --- Scheduling destruction of call '328617546726e5d430538e8061771...@192.168.250.112' in 15000 ms Found user '4053' -- SIP read from 192.168.250.112:5060: ACK sip:3...@192.168.250.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch
[asterisk-users] queue member state in asterisk 1.4
Hi, My queue members use Local channels and their queue state is In use while their hint value is Idle. Since I have Ringinuse=no, I'm experiencing issues such as incoming calls waiting too much because the agent's phone isn't ringing even though it's idle/free. I read somewhere that this is a known bug in 1.4 and should be fixed in 1.6. I think there's a backport somewhere though. Can anyone please point me to it? My scenario: queue 4000 reports agent 4002 in use when it really is idle: # asterisk -rx show queue 4000 4000 has 2 calls (max 2) in 'ringall' strategy (93s holdtime), W:0, C:479, A:127, SL:0.2% within 0s Members: Local/4...@from-internal/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/4...@from-internal/n with penalty 1 (dynamic) (In use) has taken 85 calls (last was 264 secs ago) Local/4...@from-internal/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Callers: 1. SIP/4053-5db5 (wait: 0:15, prio: 0) 2. IAX2/coinbound-5391 (wait: 0:06, prio: 0) # asterisk -rx show hints 4...@ext-local : SIP/4002Custom:DND4 State:Idle Watchers 0 - 584 hints registered # queues.conf [general] persistentmembers=yes [default] [4000] announce-frequency=75 announce-holdtime=no autofill=no eventmemberstatus=no eventwhencalled=yes joinempty=strict leavewhenempty=strict maxlen=2 music=operators periodic-announce-frequency=0 queue-callswaiting=queue-callswaiting queue-thankyou=queue-thankyou queue-thereare=queue-thereare queue-youarenext=queue-youarenext retry=0 strategy=ringall timeout=75 weight=0 wrapuptime=0 ringinuse=no -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and gnokii on same server: scratchy sound
Hi, Has anyone tried to use gnokii to send/receive SMS messages via serial or USB with AT commands while running Asterisk? Some of my calls have a scratchy sound once in a while. It doesn't seem to be due to packet loss but some kind of interference (CPU is ok, etc.). I've noticed some coincidence in time between this scratchy sound and the gnokii process. I have a bash script that calls gnokii periodically to send/receive messages. The bad audio quality does not *always* appear when the gnokii process is up but just *sometimes*. If I stop my script, thus gnokii, it seems that audio quality is fine overall. What I still don't quite understand is who's responsible for this audio problem: gnokii itself (I don't think so), the GSM radio signal nearby (about 2 meters) or the data sent through the serial port/cable. The third explanation is the most probable but I'd like to know other people's opinions. Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue members
Hi, ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053 added via -QueueAdd(4050, Local/4...@from-internal/n, 1) (not via agents.conf). SIP extension 4053 rings, answers and then decides to blind-transfer to ZAP/DAHDI extension 3666. The show queue command still displays 4053 as In use. However, if 3210 calls 4050 and 4053 answers and finally hangs up (no transfer) then the show queue command does not display In use. What's the difference? # asterisk -rx show queue 4050 -- Remote UNIX connection 4050 has 0 calls (max 6) in 'ringall' strategy (1s holdtime), W:0, C:1,A:1, SL:0.0% within 0s Members: Local/4...@from-internal/n with penalty 1 (dynamic) (In use) has taken 1 calls (last was 99 secs ago) No Callers Verbosity is at least 3 # asterisk -rx show channels concise Zap/2-1:from-alcatel-custom:s:1:Up:Bridged Call:Local/4...@from-internal-4120,2:4053::3::Local/4...@from-internal-4120,2 Local/4...@from-internal-4120,2:macro-dialout-trunk:s:19:Up:Dial:ZAP/g1/3666|300|tTwWM(auto-blkvm):3210::3:122:Zap/2-1 Local/4...@from-internal-4120,1:from-internal:s:1:Up:Bridged Call:IAX2/coinbound-1551:3210::3::IAX2/coinbound-1551 IAX2/coinbound-1551:ext-queues:4050:19:Up:Queue:4050|t||:3210::3:128:Local/4...@from-internal-4120,1 Verbosity is at least 3 Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286
--- On Fri, 4/30/10, Raimund Sacherer r...@runsolutions.com wrote: Hi, I had to choose between an 8 port FXS device from Cisco/Linksys (the sipura 3000) and a similar device from Grandstream. A look on the Grandstream's forums had me scratching my had, so much people with problems, frequently needed restarts, etc. The next thing, the Cisco/Linksys seems to be manufactured (at least this device) with durability in mind, it includes a Fan and a sturdy aluminium case, wheres the Grandstream was plastic and as far as I recall had no cooling. I have quite a few Grandstream GXW4008 devices and I must say that early firmware versions were a disaster. However, it's been at least a year now that I'm running these devices with no major problem with their latest firmware. I'm not biased and must say that they're stable now. I also have a Linksys SPA8000 (8-port ATA equivalent) with internal fan, etc., but despite its stability I've had a few non-critical issues with transfers and early dials. I must say however that support is a tad better in Grandstream than Linksys. As far as having an internal fan for cooling, I don't know if that's actually better... In general, these devices shouldn't need to rely on mechanical cooling which tends to fail in time (sure, you can open the case and replace it but that's extra maintenance). Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX trunks and audio codecs
Hi, I have IAX trunks between Asterisk servers. They receive calls on ISDN cards and Dial() through the IAX trunks to the primary Asterisk server where all the SIP phone extensions are registered. The IAX trunk settings are something like this (all servers have this identical except for the host field): [inbound] deny=all allow=alaw allow=gsm type=friend host=192.168.250.111 secret=inboundpass auth=plaintext requirecalltoken=no qualify=yes context=from-inbound username=inbound trunk=yes I'm trying to force the use of alaw because some of the local SIP extensions use this codec (a minor percentage use gsm) and none use ulaw. So I suppose that if the first Asterisk server that receives the call and sends it out to the main server via IAX encodes in alaw then the main server won't have to transcode if the destination is also alaw (most SIP phones). This should save some CPU processing in the main Asterisk server, right? So my trouble is with this message on the main Asterisk server when it receives a call from a secondary server via IAX: Apr 30 12:19:59] NOTICE[14517] channel.c: Dropping incompatible voice frame on IAX2/inbound-2255 of format alaw since our native format has changed to 0x4 (ulaw) Why is it changing to ulaw if I'm explicitly allowing only alaw and gsm and denying the rest? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping incompatible voice frame
Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] simple dialplan question
Sorry for the simple question.
I'm trying to match sipprovider.nocredit but the following doesn't execute
NoOp (it runs context but not context-custom). What am I doing wrong?
[context]
include = context-custom
exten = _.,1,Set(GROUP()=1)
exten = _.,n,Goto(destcontext,${EXTEN},1)
[context-custom]
exten = sipprovider.nocredit,1,NoOp(No credit left)
Thanks
Vieri
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[asterisk-users] sip jitter buffer
Hi, Does enabling a jitter buffer in sip.conf make sense if the call is pure SIP? SIP client---ASTERISK SIP---Internet SIP provider I think it should help on the Asterisk receiving side in case of unreliable bandwidth. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
--- On Sun, 4/25/10, Gordon Henderson gordon+aster...@drogon.net wrote: Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. If you can run ntpdate and it sets the time, then you are not running ntpd. The 2 can not run at the same time. Hi Gordon, Are you sure about this? ntpd is a daemon and adjusts the time in a continuous manner. ntp-client or ntpdate or whatever are one-time clients that reset the system clock. I don't see why an ntp-client can't be run while ntpd is working (it shouldn't be necessary but may come in handy when the time difference is big and ntpd refuses to sync). Anyway, I've noticed that my ntpd log messages don't say anything when trying to sync to my Windows PDC LAN time server. Curiously, ntp-client DOES sync to this Windows server. So I decided to sync to pool.ntp.org and now I see syslog messages that actually show that the system time gets adjusted by ntpd. I'd rather sync to my LAN time server but this is off-topic on this ML. How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Who cares. Just fix ntpd then your worys are gone. Well, I still have doubts about that. I could look at * source code but I'd rather hear from someone here. My ntp log shows this: 26 Apr 13:06:30 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:21:24 ntpd[534]: time reset +2.318647 s 26 Apr 13:21:44 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:37:46 ntpd[534]: time reset +2.325417 s 26 Apr 13:38:06 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 13:54:11 ntpd[534]: time reset +2.327974 s 26 Apr 13:55:19 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:09:16 ntpd[534]: time reset +2.177572 s 26 Apr 14:10:08 ntpd[534]: synchronized to xxx.xxx.xxx.xxx, stratum 2 26 Apr 14:26:07 ntpd[534]: time reset +2.357017 s That kind of scares me because if I'm not mistaken it means that about every 20 seconds, my ntpd adjusts the system time by about 2 seconds forward. So my clock is going back 2 seconds every 20... That's a significant drift. And it would definitely make a difference in my CDR records IF Asterisk were to compare the start and end system times. Should I worry about this? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware clock drift and CDR
while ntpd was running: # ps ax | fgrep ntp 1256 ?Ss 0:00 /usr/sbin/ntpd -p /var/run/ntpd.pid -u ntp:ntp 1623 pts/14 S+ 0:00 fgrep ntp # ntpdate -b -u pool.ntp.org 26 Apr 19:41:18 ntpdate[2791]: step time server 163.117.131.239 offset 0.142263 sec By the way, as a side question, on another server I see this: # ntpq -c peers remote refid st t when poll reach delay offset jitter == inf-srv1.hospit .LOCL. 1 u 56 64 3770.314 21755.8 7.634 Not sure what LOCL means but I'll refer to the NTP docs (inf-srv1 is my LAN Windoze time server). Anyway, back to the faulty new server (which reports a stratum of 3 after ntpd has been running for a while and sync'ing to pool.ntp.org): it's supposed to be a good motherboard (Asus) but I'm running a relatively old kernel (2.6.23). Googling around suggests me to try to boot with noapic if I keep seeing my clock drift so much. # more /proc/interrupts CPU0 CPU1 CPU2 CPU3 0:103 0 0 1 IO-APIC-edge timer 1: 2151 0 0 9 IO-APIC-edge i8042 4: 12772543 1321793296030647661766 IO-APIC-edge serial 8: 1 0 1 0 IO-APIC-edge rtc 9: 0 0 0 1 IO-APIC-fasteoi acpi 12: 0 0 0 4 IO-APIC-edge i8042 14: 2234 73664 0 2470 IO-APIC-edge ide0 16: 28322780 51914617 40744985 39615361 IO-APIC-fasteoi eth0 17: 63242610 42157366 43790794 48255583 IO-APIC-fasteoi eth1 18:1348544 0 0 1 IO-APIC-fasteoi eth2 20:9006839824429560765954923525 IO-APIC-fasteoi ahci 21: 162750903 140985080 176469550 166839225 IO-APIC-fasteoi wcte12xp0 22: 16662710 18210608 12053147 12739782 IO-APIC-fasteoi HFC-multi NMI: 0 0 0 0 LOC: 64546905 64546897 64546897 64546897 ERR: 0 MIS: 0 I have 3 PCI cards: 1 PRI, 1 quad BRI, 1 dual ethernet. Could booting with noapic help? What about my PCI devices? Will they be stable even with noapic? The reason I got this new mobo is that the previous hardware froze the system with a kernel crash. In fact, I rsync'ed to this new hardware (so identical system software) and it has been running flawlessly for more than a week now, while it used to crash/freeze once a day (another Asus board, by the way). My only problem now is with the d...@!mned clock... As far as syslog messages, I don't see anything wrong. No errors whatsoever. Thanks for your time. I'll try to boot with noapic and cross my fingers. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hardware clock drift and CDR
Hi, I've noticed that one of my new servers (new mobo) if drifting slowly backwards in time (in aprox. 24 hours, system time drifts back 5 minutes). I have an ntpd process which is supposed to sync with a lan time server but it's not quite working. So I'm launching a manual ntpdate or ntp-client once an hour and that seems to work. However, suppose I update system time at every hour and it sets +1 minute (due to a -1 minute drift). Suppose a call is dialed at 03:58 and lasts 4 real minutes. According to the updated system time, the call will have lasted 5 minutes (4+1 drift). How does Asterisk CDR count the duration/billsec values? Does it rely on system time ONLY for call start or also for call end? What Asterisk-related side-effects should I expect from a drifting clock? Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP gain
Hi, Are SIP gain parameters available in Asterisk 1.4/1.6? I'm wondering if I can increase transmission gain on SIP channels. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls drop after 20 seconds
--- On Mon, 4/19/10, Alejandro Recarey alexreca...@gmail.com wrote: their calls drop after 20 seconds or so. All of my customers use Grandstream GXW4004 telephony adapters. Check out the early dial feature in the Grandstream products (if you enabled it) and play with the pedantic option. You might want to take a look at this: https://issues.asterisk.org/view.php?id=14652 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP one-way audio
Hi, This problem has been tackled over and over, I know. I'm trying to understand why I'm having trouble with my simple setup. My setup is like this: SIP_PROVIDER---DSL1---LINUX_GATEWAY---ASTERISK_VIA_DSL1 I've unloaded the nf_*_sip kernel modules from the LINUX_GATEWAY just in case they could interfere. The DSL1 modem/router is a THOMSON SPEEDTOUCH and I've disabled SIP ALG. If I place a SIP call then I immediately get a one-way audio issue. However, if I define externip and localnet in my sip.conf and sip reload then at first, calls are OK (two-way) for about 10-15 minutes. After that, all calls are one-way again... If I remove the externip and localnet settings in my sip.conf and sip reload, then re-enable SIP ALG in the DSL1 modem, then calls are two-way for about 10-15 minutes. After that, all calls are one-way again... Is it somehow timing out? By the way, no firewall incoming natting rules are defined in the Linux gateway. I'm using the internet SIP provider for outgoing calls only. Any ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cause 66 - Channel not implemented
Hi, What can I make of the following log messages? Extension 7114 tries to reach 6035 but gets an unknown channel type. What does it mean? (supposedly, 6035 was not busy...) Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing Dial(SIP/7114-b4fe1ef0, /6035|300|) in new stack Apr 12 13:01:01 WARNING[30989] channel.c: No channel type registered for '' Apr 12 13:01:01 NOTICE[30989] app_dial.c: Unable to create channel of type '' (cause 66 - Channel not implemented) Apr 12 13:01:01 VERBOSE[30989] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Thanks Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] scratchy sound
Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to the bandwidth between my peer and the SIP provider). You can hear a scratchy sound during the whole fragment. I can't determine the possible cause of this kind of distortion. Maybe an expert ear can give me a clue. It shouldn't be the Asterisk server's CPU usage. Could it be a network issue between the Asterisk system and the SIP client? (it happens with SIP hardphones as well as softphones so I guess it's improbable it's the client software/firmware) Both softphones and hardphones use GSM and usually work fine (this kind of issue is not too frequent). The LAN isn't dedicated to voice but has QoS prioritizing VoIP. Could the cause of the distortion be network-related? And only on my side? Should I consider other causes? Thanks, Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crash - segmentation fault
--- On Tue, 3/23/10, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote:
--- On Tue, 3/23/10, Vieri rentor...@yahoo.com
wrote:
My Asterisk 1.2.40 process crashes
regularly in the is_zero_or_null function at:
return (*vp-u.s == 0 || (to_integer (vp)
vp-u.i == 0));
My gdb trace is at:
http://pastebin.com/raw.php?i=hmhzZxye
Other examples here:
http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html
Can anyone please help?
And my Asterisk log shows the following right before
the crash:
Mar 23 12:32:37 VERBOSE[9054] logger.c:
-- Executing
ExecIf(SIP/4070-09464648,
0|Set|REALCALLERIDNUM=4070) in new stac k
Mar 23 12:32:37 DEBUG[9054] app_macro.c: Executed
application: ExecIf
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '0'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Function result is
'4070'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '0'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '1'
Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result
is '0'
Mar 23 12:32:37 WARNING[9054] ast_expr2.y: Conversion
of 0 to integer
under/overflowed!
What does this mean?
It's quite clearly a bug, but given that 1.2 is in security
maintenance mode,
it's not a bug that will ever be fixed in an official
release of Asterisk.
Your best bet is to bite the bullet and upgrade to 1.4.
Understood. However, 1.4 also has the same code for that function.
There's something I'd like to know about this logic:
errno = 0;
i = strtoll(vp-u.s, (char**)NULL, 10);
if (errno != 0) {
ast_log(LOG_WARNING,Conversion of %s to integer
under/overflowed!\n, vp-u.s);
free(vp-u.s);
vp-u.s = 0;
return(0);
}
Since my warning message is Conversion of 0 to integer under/overflowed! then
that means the string was set to 0 before the conversion.
0 is within the range LLONG_MIN - LLONG_MAX.
So what I don't understand is why strtoll is failing if vp-u.s is actually 0.
Wouldn't that fail in 1.4 too?
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[asterisk-users] Asterisk crash - segmentation fault
My Asterisk 1.2.40 process crashes regularly in the is_zero_or_null function at: return (*vp-u.s == 0 || (to_integer (vp) vp-u.i == 0)); My gdb trace is at: http://pastebin.com/raw.php?i=hmhzZxye Other examples here: http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html Can anyone please help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crash - segmentation fault
--- On Tue, 3/23/10, Vieri rentor...@yahoo.com wrote: My Asterisk 1.2.40 process crashes regularly in the is_zero_or_null function at: return (*vp-u.s == 0 || (to_integer (vp) vp-u.i == 0)); My gdb trace is at: http://pastebin.com/raw.php?i=hmhzZxye Other examples here: http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html Can anyone please help? And my Asterisk log shows the following right before the crash: Mar 23 12:32:37 VERBOSE[9054] logger.c: -- Executing ExecIf(SIP/4070-09464648, 0|Set|REALCALLERIDNUM=4070) in new stac k Mar 23 12:32:37 DEBUG[9054] app_macro.c: Executed application: ExecIf Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 DEBUG[9054] pbx.c: Function result is '4070' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '1' Mar 23 12:32:37 DEBUG[9054] pbx.c: Expression result is '0' Mar 23 12:32:37 WARNING[9054] ast_expr2.y: Conversion of 0 to integer under/overflowed! What does this mean? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 crash: gdb trace on core dump
Hi, I'm one of those people who still need to maintain * 1.2 systems and cannot easily upgrade. :-( My 1.2 systems were very stable until I upgraded from 1.2.37 to 1.2.40. I have made some changes within my dialplan but nothing unusual. Today I've had a crash: https://issues.asterisk.org/file_download.php?file_id=25571type=bug Yesterday I had another: https://issues.asterisk.org/file_download.php?file_id=25572type=bug Could anyone please have a look at these gdb traces? Other than that the traces seem to point to ast_expr2 and chan_iax2, I don't really have a clue as to why Asterisk crashes. Any ideas? Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
