[Asterisk-Users] Re: Problem with ADIT 600 and FXO configuration
Hello, OK, some things I've found out so far. The ground connection to the ADIT chassis wasn't really to ground (fixed that, it made FXS card happy when connected). Taking a cue from another post I also reduced the number of options specified in zapata.conf to: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes group=1 callgroup=1 pickupgroup=1-2 immediate=no musiconhold=default group = 0 signalling=fxs_ks context = incoming busydetect = no overlapdial = no channel = 25-27 signalling=fxs_ks channel = 97 ;X100P group = 1 signalling = fxo_ks context = internal channel = 98-100 channel = 101-105 Using zttool I tried to loopback the TE406P span 1 which switched the ADIT a:2 port into loop back, setting the line down and back up didn't clear the configuration (I had to find the set a:2 line loopdown command). Moving the link to span 2 on the TE406P I now can receive incoming calls (yea!), trying to place an outbound call results in dead air with the eventual message that the call didn't go through :-( Note that both the ADIT and the TE406P were showing green on the T1 connection however it wasn't until I changed the connection to span 2 that it started allowing inbound calls to work, zap show channel 1 showed InAlarm: 1 although I didn't spot any other error messages. zztool currently shows: RED/NOP T4XXP (PCI) Card 0 Span 1 OK T4XXP (PCI) Card 0 Span 2 RED T4XXP (PCI) Card 0 Span 3 RED T4XXP (PCI) Card 0 Span 4 RED Wildcard X101P Board 1 OK Wildcard TDM400P REV E/F Board 1 OK Wildcard TDM400P REV E/F Board 2 The NOP on Span 1 appears to mean Not Opened however I don't know what that means. I've got one more day/night to get this working so any suggestions are welcome. Thank you, William. On Mon, 2005-11-28 at 03:28, William K. Volkman wrote: I've looked through the archives of the mailing list for the last year and although informative I've not been successful at get this to work. We had a working Asterisk PBX system with 3 Digium X101P FXO lines and two TDM400P FXS cards. I've setup an ADIT 600 with an 8 port FXO card (and an 8 port FXS card not currently installed). We are going to be adding a T1 for incoming calls this week. I removed two of the X101P cards and installed a TE406P. I'm using Asterisk 1.0.9 (and matching zaptel, libpri) from tar files. /etc/zaptel.conf has this configuration: span=1,1,0,esf,b8zs,yellow span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs #Modular unit, first card is FXO fxsks=1-3 unused=4-8 #Modular unit, 1 FXS cards unused=9-16 unused=17-24 unused=25-48,49-72,73-96 fxsks=97 fxoks=98-101 fxoks=102-105 /etc/asterisk/zapata.conf has this: group = 0 signalling=fxs_ks context = incoming busydetect = yes overlapdial = no channel = 1-3 signalling=fxs_ks channel = 97 ;X100P group = 1 signalling = fxo_ks context = internal ;TDM400P callerid = Available 200 channel = 98-100 callerid = x channel = 101 ;TDM400P callerid = x channel = 102 callerid = x channel = 103 Parts of my adit configuration: -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 4 set a:1 framing esf set a:1 id Inbound set a:1 linecode b8zs set a:1 loopdetect csu set a:1:1-24 side drop set a:1:1-24 type voice set a:1:1-24 signal ls set a:2 up set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id Outbound PBX set a:2 linecode b8zs set a:2 loopdetect csu set a:2:1-24 side drop set a:2:1-24 type voice set a:2:1-24 signal ls -Setting slot 1. set 1:1-8 signal lscpd set 1:1-8 txgain -3 set 1:1-8 rxgain -6 -Setting primary and secondary clock sources. set clock1 a:1 set clock2 internal -Setting the system idle pattern for DS0s. set idle 0xff -Making connections. connect a:2:1-3 1:1-3 Inbound calls just ring and ring (the leds on the ADIT change state) however asterisk doesn't respond. Attempts to make outgoing calls get: -- Executing Dial(SIP/202-ba07, Zap/g0/5551212) in new stack Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Congestion(SIP/202-ba07, ) in new stack == Spawn extension (from-sip, 95942060, 3) exited non-zero on 'SIP/202-ba07' -- Executing Hangup(SIP/202-ba07, ) in new stack I've tried just about all combinations of gs/ls/ks for the signalling to no avail. Here is the output of status: status a:2:1-3 DS0 Rx AB Tx AB Signal T1 TP --- - - -- - -- a:2:1 01 01
[Asterisk-Users] Problem with Ext calling
Hi all I have installed Astrix on FC4 and running successfully and installed Astbill on top of the server and able to mange accounts i have made 2 extenstions 17612 17349 and iam able to use soft SJPhone and able to register and when i try to call 17349 i get an error Address incomplete call rejected: 484 address incomplete Sip debug iam getting this error Looking for y.y.y.y :5060 in default (domain )Transmitting (NAT) to x.x.x.30:5060:SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.9.3;branch=z9hG4bKc0a809030010438c132f79d401a1;received= x.x.x.30;rport=5060From: sip:[EMAIL PROTECTED];tag=3527616543877To: sip:y.y.y.y:5060;tag=as63a97f53Call-ID: [EMAIL PROTECTED]CSeq: 210 OPTIONSUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:y.y.y.y Accept: application/sdpContent-Length: 0 any help will be aprriciated ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP rapid INVITE re-transmission: bug, or config problem?
John Todd wrote: I'm having a problem with Asterisk sending too many INVITEs to a peer for a single call. I can't quite figure out why there are these rapid INVITEs sent to the call proxy. The call completes correctly (to, in this example, an echo test found via ENUM) but the number of INVITEs is really out of control and is a Bad Thing overall. My notes: 1) This isn't a firewall problem - there aren't any. Additionally, you'll note that the INVITEs are all being replied to eventually. 2) The intervals between the INIVTEs after the 407 sequence are: 34ms, 30ms, 49ms, 91ms. This is _way_ too fast for response timers to be expiring for reliable re-transmissions of INVITEs... isn't it? According to DEFAULT_RETRANS in chan_sip.c, the proper delay should be 1000ms between retransmissions. No, it's twice the known roundtrip time or twice 500 ms. Since you have qualify on, we propably know the roundtrip time and do the retransmissions based on that. 3) Here is the peer definition for this system: [testbed] type=peer username=9 secret=dio0sywa82a host=10.0.3.173 insecure=very context=default qualify=4000 6) The INVITEs create a huge logjam of 100 Trying and 200 OK with SDP messages. This is Bad. Can we see a SIP debug? cookies*CLI testbed -- Executing Dial(SIP/2598-dbb4, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/testbed-6b7c answered SIP/2598-dbb4 -- Attempting native bridge of SIP/2598-dbb4 and SIP/testbed-6b7c -- Executing Hangup(SIP/2598-dbb4, ) in new stack cookies*CLI Looks fine. 10) The (post-407) INVITEs are identical to each other - there are no differences in Call-ID, branch, tag, nonces, or SDP. I then compared the INVITES between a working peer and the broken peer to each other - they're almost identical except for IP addresses, so nothing obvious there, either. Good. 11) Each INVITE in a sip debug output is tagged with something like Retransmitting #4 (no NAT) to 10.0.3.173:5060: but there are no timer statements that I could see in the debug. Turn on debug to 4. There should be messages about changing T1 timers then. I am _totally_ stumped here. I have changed the names of the peers, changed the qualify= statement, moved the peers around in sip.conf, stood on my head, etc. Your insights on this would be appreciated, since I'm not quite sure what Asterisk is up to with these rapid INVITE sequences. I'm thinking bug but maybe there is some subtle config option that I'm overlooking, so I'll ask the list before I file the bug. Maybe it's just that I've had too much caffeine today and the obvious solution is the one that's the most difficult to see. If you turn off qualify, we will use the default 500 ms as Timer T1. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
Leif Neland wrote: On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote: From memory (at a previous installation) you will need a newer version of Asterisk than 1.09 for the lights to work. on 1.0.9 the lights work. In this way: person is on the phone: light is on Person is not on the phone: light is off since 1.2 the lights will blink when the phone is running and above states work the same. Running? Is that a 3. state? No, a typo. If the extension is ringing the led blinks, now all we need is a way to pick up that ringing channel. Could anyone tell me where the patch is that added hint support for local channels as i need to use the led for Agents (because people here don't use a fixed desk) Erik Leif ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with ADIT 600 and FXO configuration
What does the TE406 leds indicate? Both the ADIT 600 led and the TE406 led are green, the ADIT has zeros in the error counters. Syslog has this as a final message after running ztcfg: Nov 28 02:31:08 xxx kernel: Registered tone zone 0 (United States / North America) Nov 28 02:36:21 xxx kernel: wct4xxp: Clearing yellow alarm on span 1 I've seen documentation that says that telco-pots lines use loop start and I've seen mailing list entries that says you should use ground start for reliability. Can anyone clarify this? Back in the olden analog days, loop start trunks had an issue when calls were simultanously started at each end of the trunk. There was nothing built into the loopstart mechanism to resolve which end got the trunk. As a result, two unknown callers would be tied together, both complaining of wrong numbers. Ground start trunks was a solution to that analog problem, thus making them more reliable. Other ways to make loopstart trunks more reliable included have one end of the trunks always start using trunks from the low numbered end (eg, 1, 2, 3), and the opposite end start with high numbered trunks (eg, 24, 23, 22). The asterisk implementation of that is g1 and G1 for zap channel order selection. In lightly loaded systems the above is generally not a problem. On heavily loaded systems with a reasonable mix of incoming and outgoing trunk calls, loopstart trunks can be a slight problem that is most often addressed through the trunk selection mechanism (eg, g1 vs G1). Pure guess is the ground start functionality was implemented in asterisk due to interface requirements to some legacy systems, and not as a workaround for loopstart issues. If your ADIT 600 has fxo cards in it, the selection of analog loop start vs ground start will likely be dictated by whatever box or central office switch your connecting the analog wires to. Loop start is by far more common in todays telephony environment. Pots lines are always loop start in the US. Keep in mind that in the analog days, trunks were implemented with a series of relays (and other electromechanical devices), and there was little that one could do in terms of controlling signal timing. That timing could range from 100 milliseconds to as much as a second or so depending upon exactly what equipment was used. With T1/E1's, signaling happens in a few milliseconds and does not represent the same problem magnitude. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digitmap problems
I'm trying to implement some of the star services such as *61 for weather or *71 for wakeup call, etc. I think I have asterisk setup properly because I can get them to work fine using normal extension numbers. However, if I try to use the 'star' numbers, my Polycom IP500 never sends the digits to asterisk, at least I never see Asterisk try to do anything in the logs. I believe the phone is giving me a fast busy signal because it can not find a match in the digitmap. I've tried digitmaps like: *6x|*7x|2xxx|[2-9]x What am I missing??? The Admin Guide? I searched through the v1.5 guide, and it implies the digitmap uses numbers only (no * or #). But, it doesn't actually discuss it either. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM Phones MWI, Hold Retrieve buttons notworking with Asterisk v1.2
If you edit sip.conf in 1.2 and put
Vmexten = voicemail
Fromdomain = yourip or domain of the asterisk box
Then in extensions.conf
exten = voicemail,1,VoicemailMain(${CALLERIDNUM})
That works and look nicer on the snom phones.(it dials voicemail)
Under 1.2 you can put in sip.conf
Fromuser = voicemail
But this effects everything not just the voicemail
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sascha
Deri
Sent: 28 November 2005 23:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SNOM Phones MWI, Hold Retrieve buttons
notworking with Asterisk v1.2
I made an error in what I previously wrote. What actually works in v1.2
is:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
Which is what Michael originally wrote. My bad!
Sascha wrote:
Thanks Michael - you got me on the right path. What you gave me didn't
work, but I figured out that the following does (on version 1.2):
exten = default,1,VoiceMailMain(${CALLERIDNUM})
(BTW, exten = Unknown,1,VoiceMailMain(${CALLERIDNUM}) used to work
for us in Asterisk 1.0.9 but obviously no longer does)
___
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[Asterisk-Users] Hangup after 18 sec on PRI channel
Hi I have a Te411 PRI card connected to parlay voxtream i60. Every call that comes on asterisk over zap channel and goes on to SIP Voice Blue gsm gateway disconects after this timeout. This is complete sip debug log. I also described how sip communication is done in this matter. My configuration for sip is very simple i have a trunk number 5 called gsm_gw_1_1-peer with following settings. Voice Blue is ip gsm gateway and it is working ok on several instalations but never with PRI card. This disconnect happens because calling equipment doesn't get any response from Asterisk on zap channel that call is in progress. Why aren't message from sip forwarded to zap channel? Would it be better if Ringing message would be sent from voice blue instead of session progress? [general] canreinvite=no bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=alaw [gsm_gw_1_1-peer] type=peer host=192.168.0.100 dtmfmode=inband context=from-mux canreinvite=no Asterisk PBX VoiceBlue INVITE TRYING SESSION PROGRESS CANCEL OK REQUEST TERMINATED ACK - Starting simple switch on 'Zap/3-1' -- Accepting overlap call from '38626540259' to '041656699' on channel 0/3, span 1 -- Executing Goto(Zap/3-1, outrt-005-IpGsmGateway13|0038641656699|1) in new stack -- Goto (outrt-005-IpGsmGateway13,0038641656699,1) -- Executing Macro(Zap/3-1, dialout-trunk|5|0038641656699|) in new stack -- Executing GotoIf(Zap/3-1, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(Zap/3-1, user-callerid) in new stack -- Executing DBget(Zap/3-1, AMPUSER=DEVICE/38626540259/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=38626540259/user -- DBget: Value not found in database. -- Executing DBget(Zap/3-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(Zap/3-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(Zap/3-1, Using CallerID 38626540259) in new stack -- Executing Macro(Zap/3-1, record-enable|38626540259|OUT) in new stack -- Executing GotoIf(Zap/3-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/3-1, recordingcheck|20051129-095434|1133254470.611) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20051129-095434|1133254470.611: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/3-1, No recording needed) in new stack -- Executing Macro(Zap/3-1, outbound-callerid|5) in new stack -- Executing GotoIf(Zap/3-1, 1?3) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing DBget(Zap/3-1, USEROUTCID=AMPUSER/38626540259/outboundcid) in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=38626540259/outboundcid -- DBget: Value not found in database. -- Executing GotoIf(Zap/3-1, 1?6) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp(Zap/3-1, CallerID set to 38626540259) in new stack -- Executing SetGroup(Zap/3-1, OUT_5) in new stack -- Executing CheckGroup(Zap/3-1, ) in new stack -- Executing SetVar(Zap/3-1, DIAL_NUMBER=0038641656699) in new stack -- Executing SetVar(Zap/3-1, DIAL_TRUNK=5) in new stack -- Executing AGI(Zap/3-1, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(Zap/3-1, OUTNUM=10038641656699) in new stack -- Executing Cut(Zap/3-1, custom=OUT_5|:|1) in new stack -- Executing GotoIf(Zap/3-1, 0?16) in new stack -- Executing Dial(Zap/3-1, SIP/gsm_gw_1_1-peer/10038641656699) in new stack We're at 192.168.0.99 port 13554 Adding codec 0x8 (alaw) to SDP 13 headers, 8 lines Reliably Transmitting (no NAT) to 192.168.0.100:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport From: 38626540259 sip:[EMAIL PROTECTED];tag=as6809c997 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 29 Nov 2005 08:54:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp
Re: [Asterisk-Users] Re: Problem with ADIT 600 and FXO configuration
Well... I don't have an ADIT box around, so can't help on that. Do take a close look at the channel assignment stuff, both in zaptel.conf and zapata.conf. Are you absolutely sure the ordering of the cards and channels are right (haven't moved any cards around or removed any)? Your statement it wasn't until I changed the connection to span 2 that it started allowing inbound calls to work suggests the ordering of the channels might not be what you are expecting. You have channels 25-27 defined in zapata.conf, but they are shown as unused in zaptel.conf. (I did not try to match up all the other ones.) Take a close look at the group= definitions below. First set to group=1, then six lines below that its group=0. Are you calling out with an extensions.conf entry like Zap/g1? And, are all the channels that are included in g1 actually connected/usable? (eg, be carefull with assumptions about what happens when a channel is included in the group definition but the associated ADIT port isn't connected to anything.) Instead of using Zap/g1, prove to yourself things are configured correctly by sending calls to Zap/99 (or whichever channel you have connected to a real line), and do that for each fxo line that you think is wired/working. Might look at 'zap show status' and 'zap show channels' to ensure what your expecting is what is defined. RED/NOP: RED generally means the T1 port is not seeing any timing signals (eg, nothing is connected to it). NOP generally mean Not-OPerational. Not sure why T1 port #1 on the card didn't work. Could be a bad port or the channel #'s aren't as you expect. You can test for a bad port by creating a T1 crossover cable, and send test calls out one T1 and receive those calls on another T1 (on the same card). Last, any changes made to zapata.conf requires a complete restart of asterisk (not just a reload). And, any changes to zaptel.conf requires a reload of the zaptel drivers. Rich Hello, OK, some things I've found out so far. The ground connection to the ADIT chassis wasn't really to ground (fixed that, it made FXS card happy when connected). Taking a cue from another post I also reduced the number of options specified in zapata.conf to: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes group=1 callgroup=1 pickupgroup=1-2 immediate=no musiconhold=default group = 0 signalling=fxs_ks context = incoming busydetect = no overlapdial = no channel = 25-27 signalling=fxs_ks channel = 97 ;X100P group = 1 signalling = fxo_ks context = internal channel = 98-100 channel = 101-105 Using zttool I tried to loopback the TE406P span 1 which switched the ADIT a:2 port into loop back, setting the line down and back up didn't clear the configuration (I had to find the set a:2 line loopdown command). Moving the link to span 2 on the TE406P I now can receive incoming calls (yea!), trying to place an outbound call results in dead air with the eventual message that the call didn't go through :-( Note that both the ADIT and the TE406P were showing green on the T1 connection however it wasn't until I changed the connection to span 2 that it started allowing inbound calls to work, zap show channel 1 showed InAlarm: 1 although I didn't spot any other error messages. zztool currently shows: RED/NOP T4XXP (PCI) Card 0 Span 1 OK T4XXP (PCI) Card 0 Span 2 RED T4XXP (PCI) Card 0 Span 3 RED T4XXP (PCI) Card 0 Span 4 RED Wildcard X101P Board 1 OK Wildcard TDM400P REV E/F Board 1 OK Wildcard TDM400P REV E/F Board 2 The NOP on Span 1 appears to mean Not Opened however I don't know what that means. I've got one more day/night to get this working so any suggestions are welcome. Thank you, William. On Mon, 2005-11-28 at 03:28, William K. Volkman wrote: I've looked through the archives of the mailing list for the last year and although informative I've not been successful at get this to work. We had a working Asterisk PBX system with 3 Digium X101P FXO lines and two TDM400P FXS cards. I've setup an ADIT 600 with an 8 port FXO card (and an 8 port FXS card not currently installed). We are going to be adding a T1 for incoming calls this week. I removed two of the X101P cards and installed a TE406P. I'm using Asterisk 1.0.9 (and matching zaptel, libpri) from tar files. /etc/zaptel.conf has this configuration: span=1,1,0,esf,b8zs,yellow span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs #Modular unit, first card is FXO fxsks=1-3 unused=4-8 #Modular unit, 1 FXS cards unused=9-16 unused=17-24 unused=25-48,49-72,73-96 fxsks=97 fxoks=98-101 fxoks=102-105 /etc/asterisk/zapata.conf has this: group = 0 signalling=fxs_ks context = incoming busydetect = yes overlapdial = no channel = 1-3 signalling=fxs_ks channel =
[Asterisk-Users] Re: SNOM and 1.0.9
I still cannot get this to work on 1.0.9. I am trying to test with two extensions: Here is the config I am using: exten = 451,hint,sip/451 exten = 451,1,Dial(SIP/451,20,tr) exten = 451,2,Voicemail([EMAIL PROTECTED]) exten = 451,102,Voicemail([EMAIL PROTECTED]) exten = 453,hint,sip/453 exten = 453,1,Dial(SIP/453,20,tr) exten = 453,2,Voicemail([EMAIL PROTECTED]) exten = 453,102,Voicemail([EMAIL PROTECTED]) On the SNOM, the SIP trace shows the initial subscription: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK13ea176f From: sip:[EMAIL PROTECTED];user=phone;tag=as77402d3b To: sip:[EMAIL PROTECTED];tag=c0av8f2x4v Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 203 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0 state=full entity=sip:[EMAIL PROTECTED] dialog id=453 stateterminated/state /dialog /dialog-info The SNOM shows the light off for this extension. This is a hardphone, and is always registered. NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 195.27.242.8:5060;branch=z9hG4bK258fb569 From: sip:[EMAIL PROTECTED];user=phone;tag=as26ba79ca To: sip:[EMAIL PROTECTED];tag=8ioo4i3sp7 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 202 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0 state=full entity=sip:[EMAIL PROTECTED] dialog id=451 stateconfirmed/state /dialog /dialog-info This is a softphone that is not registered, and the light on the keyboard is on. Light is one unavailable, light is off available. When I make a call from extension 453, and am on the phone, nothing is sent to the SNOM. I see no SIP packets leaving Asterisk either. This is what Asterisk shows: asterisk_test*CLI sip show subscriptions Peer UserCall IDURI 195.27.242.113 320 3c26700c30d4-libo7sf1 195.27.242.113 320 3c26700c2bf2-wfqpeg34 0 active SIP subscriptions(s) asterisk_test*CLI If anyone has any additional ideas, or a snippet of config that works, please post it. I will try to upgrade to 1.2 and see how this works. Thanks, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comedian Voicemail? PROBLEMS?
On Nov 28, 2005, at 3:55 PM, BJ Weschke wrote: On 11/28/05, Martin Joseph [EMAIL PROTECTED] wrote: snipI am only able to get comedian voicemail (ie dialing 1234) to record or playback messages if I use the GSM codec? Is this normal and expected? If I use ulaw or alaw I get either trash noise or an immediate busy signal on attempted message playback. I am running asterisk 1.2 on OSX 10.4.3. snip This is definitely not normal or expected. Are there any errors that come up on the CLI? snip It seems to be running along smoothly until it attempts playback and then... Nov 29 02:22:35 WARNING[38]: format_wav.c:153 check_header: Not a wav file 49 Nov 29 02:22:35 WARNING[38]: file.c:432 ast_filehelper: Unable to open file on /var/spool/asterisk/voicemail/default/1234/Old/msg.wav Nov 29 02:22:35 WARNING[38]: file.c:820 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/1234/Old/msg (format alaw): No such file or directory == Spawn extension (autocontext, 8500, 1) exited non-zero on 'IAX2/2001-2' -- Hungup 'IAX2/2001-2' I do appreciate the attention and hopefully helpful suggestions? Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pros and Cons of T1/E1 cards
Hi Luke, It's important to compare apples and pears though. The card you mentioned has 24 on board Digital Signal Processors that enable it to do the following: Tone Detection Voice Activity Detection Conferencing with automatic Gain Control and echo cancellation Continuous full duplex audio support Speech recognition support This means for example that the card could be used for a conferencing application with 24 users with echo cancellation/ gain control being handled by the card - and not having to be processed by the central CPU. Full information can be found here: http://tinyurl.com/dnphn I hope this clarifies things. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problem with ADIT 600 and FXO configuration
Hello, On Tue, 2005-11-29 at 02:25, Rich Adamson wrote: Well... I don't have an ADIT box around, so can't help on that. Do take a close look at the channel assignment stuff, both in zaptel.conf and zapata.conf. Are you absolutely sure the ordering of the cards and channels are right (haven't moved any cards around or removed any)? Your statement it wasn't until I changed the connection to span 2 that it started allowing inbound calls to work suggests the ordering of the channels might not be what you are expecting. You have channels 25-27 defined in zapata.conf, but they are shown as unused in zaptel.conf. (I did not try to match up all the other ones.) Sorry, I had also make the requisite changes in zaptel.conf: span=1,0,0,esf,b8zs,yellow span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs fxsks=1-8 unused=9-16 unused=17-24 fxsks=25-48 unused=49-72,73-96 fxsks=97 fxoks=98-101 fxoks=102-105 loadzone = us defaultzone=us Take a close look at the group= definitions below. First set to group=1, then six lines below that its group=0. Are you calling out with an extensions.conf entry like Zap/g1? And, are all the channels that are included in g1 actually connected/usable? (eg, be carefull with assumptions about what happens when a channel is included in the group definition but the associated ADIT port isn't connected to anything.) Instead of using Zap/g1, prove to yourself things are configured correctly by sending calls to Zap/99 (or whichever channel you have connected to a real line), and do that for each fxo line that you think is wired/working. Yes the calls out are/were to Zap/g1/xxx, changing them to the specific Zap channels makes no difference. I just now tried adding w to the dial stream, no effect. Discovered that my new test-set shows DTMF digits, hooked it up and I'm seeing only the first digit of the phone number being sent on the outgoing line (the reason for the Call didn't go through message). Any ideas where next to look? Might look at 'zap show status' and 'zap show channels' to ensure what your expecting is what is defined. Is show status a asterisk 1.2 command? *CLI zap show status No such command 'zap show status' (type 'help' for help) *CLI zap show channels Chan Extension Context Language MusicOnHold pseudointernal default 25incoming default 26incoming default 27incoming default 97incoming default 98internal default 99internal default 100internal default 101internal default 102internal default 103internal default 104internal default 105internal default *CLI RED/NOP: RED generally means the T1 port is not seeing any timing signals (eg, nothing is connected to it). NOP generally mean Not-OPerational. When the cable is connected to span1 the RED goes away but it stays in NOP. Not sure why T1 port #1 on the card didn't work. Could be a bad port or the channel #'s aren't as you expect. You can test for a bad port by creating a T1 crossover cable, and send test calls out one T1 and receive those calls on another T1 (on the same card). I may try this tomorrow, I've got about another 1/2 hour before I have to revert the system to original/working configuration. Last, any changes made to zapata.conf requires a complete restart of asterisk (not just a reload). That I knew and have been doing... And, any changes to zaptel.conf requires a reload of the zaptel drivers. I thought that running ztcfg was sufficient. In any case I've got scripts that rmmod the modules and modprobe them before starting up asterisk. Rich Thanks btw. for that informative explanation of the loopstart v.s. groundstart signalling, could I suggest that information would be useful on the voip-info.org wiki (If not already there, I found some useful information tucked in unrelated topics). And I'm in the US using Qwest POTS lines so loopstart it is. Hello, OK, some things I've found out so far. The ground connection to the ADIT chassis wasn't really to ground (fixed that, it made FXS card happy when connected). Taking a cue from another post I also reduced the number of options specified in zapata.conf to: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes group=1 callgroup=1 pickupgroup=1-2 immediate=no musiconhold=default group = 0 signalling=fxs_ks context = incoming busydetect = no overlapdial = no channel = 25-27 signalling=fxs_ks channel = 97 ;X100P group = 1
[Asterisk-Users] setting variables in a .call file - how?
How can I set a variable in a .call file? I wanted to add a fax header with SpanDSP / txfax, and the information on soft-switch.org says: If the variable LOCALHEADERINFO has been set when txfax is run, the value of that variable will be used as the user defined part of the header text. So I tried to set that variale in a .call file: Channel: $CHANNEL/$FAXNUM MaxRetries: 2 retryTime: 60 WaitTime: 20 SetVar: LOCALHEADERINFO=CompanyName Application: txfax Data: $DATADIR/$ATTNAME.tif|caller but it doesn't make any difference, fax header is not added. So perhaps I'm setting that variable in a wrong way? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] isdn
I'm testing asteriskathome with an ISDN card 00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) I found there is the module hisax and I loaded it: hisax 456177 0 crc_ccitt 2113 2 hisax,zaptel isdn 133409 1 hisax dmesg shows this: HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 HiSax: LinkLayer Revision 2.59.2.4 I'm not sure if it is detecting the hardware, and I'm not sure what config I must do in asterisk. The documentation is confusing, because the references to hisax indicates to use cahan_modem_i4l but comments in modules.conf says DON'T load the chan_modem.so, as they are obsolete in * 1.2. I tryed anyway but chan_mdem_i4l does not appear whan I type reload. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
Alejandro Vargas schrieb: I'm testing asteriskathome with an ISDN card 00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) I found there is the module hisax and I loaded it: hisax 456177 0 crc_ccitt 2113 2 hisax,zaptel isdn 133409 1 hisax dmesg shows this: HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 HiSax: LinkLayer Revision 2.59.2.4 I'm not sure if it is detecting the hardware, and I'm not sure what config I must do in asterisk. The documentation is confusing, because the references to hisax indicates to use cahan_modem_i4l but comments in modules.conf says DON'T load the chan_modem.so, as they are obsolete in * 1.2. I tryed anyway but chan_mdem_i4l does not appear whan I type reload. you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card, not HiSax (well, technically, you could use HiSax too, but avoid that if possible). -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Load spikes with 1.0.10
Hi, I have a trivial setup on a 2.4GHz Xeon Dell PE 1750 SCSI machine
dealing with 4 ports of E1 in an 'inline PBX' arrangement.
My extensions.conf is simply:
[general]
static=yes
writeprotect=yes
[frompstn]
exten = _31.,1,Dial(Zap/g2/${EXTEN})
exten = _31.,2,Congestion
[fromaxxess]
exten = _13.,1,Dial(SIP/${EXTEN},,h)
exten = _13.,2,Congestion
exten = _31.,1,Dial(Zap/g2/${EXTEN})
exten = _31.,2,Congestion
include = outbound
[outbound]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
We have a full 30-channel PRI and a 4-channel partial PRI and are
experiencing load spikes that I can't find the source of.
The machine Debian sarge on the default 2.6.8-2-686 kernel, and no other
daemons are running than sshd.
The machine is doing no IP work - purely TDM, yet on a Xeon 2.4GHz
machine, the load average is sitting at 0.6 with 40 active Zap channels
(i.e. 20 live calls) and will randomly jump to 2 (with call quality
starting to stutter)
A few seconds of vmstat:
procs ---memory-- ---swap-- -io --system--
cpu
r b swpd free buff cache si sobibo incs us sy
id wa
7 0 0 223560 1276 22304000 310 8394 1 2
97 0
0 0 0 223552 1284 22304000 016 5128 3461 1 0
98 1
0 0 0 223552 1284 22304000 0 0 5094 3319 10 9
81 0
0 0 0 223552 1284 22304000 016 5130 2955 1 10
89 0
0 0 0 223552 1292 22304000 060 5121 2918 0 1
97 2
0 0 0 223552 1292 22304000 0 0 5031 2936 1 0
99 0
Does this sound about normal for what is just shuffling data between
ports of the Sangoma A104? I want to record the call data with the
'Monitor' application but this just causes the load to increase even
more (even though 'hdparm' shows 70MB/sec disk transfer with low
user+system CPU usage)
/proc/interrupts is
CPU0
0: 423253622IO-APIC-edge timer
1:175IO-APIC-edge i8042
9: 0 IO-APIC-level acpi
11: 0 IO-APIC-level ohci_hcd
12: 58IO-APIC-edge i8042
15: 13IO-APIC-edge ide1
177: 50 IO-APIC-level ioc0
185: 29 IO-APIC-level ioc1
193: 1311243931 IO-APIC-level wanpipe1, wanpipe2, wanpipe3, wanpipe4
201: 13289965 IO-APIC-level eth0
217:5420038 IO-APIC-level eth2
NMI: 0
LOC: 423311408
ERR: 0
MIS: 0
Help! :)
Cheers,
Gavin.
___
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[Asterisk-Users] TDM400 revisions problem: Rev J not working!!
Hi, I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm using a K8N-E deluxe asus motherboard which gives me some problems (but I'm not sure is the motherboard causing the problem): - if I plug a TDM400 REV J, Debian cannot recognize it - if I plug a TDM400 REV E/F, everything goes well Is there anybody out there who can help me?? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 revisions problem: Rev J not working!!
sure? have you tried latest drivers? could be simply a pci-id problem. matteo. Il giorno mar, 29/11/2005 alle 11.59 +0100, gincantalupo ha scritto: Hi, I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm using a K8N-E deluxe asus motherboard which gives me some problems (but I'm not sure is the motherboard causing the problem): - if I plug a TDM400 REV J, Debian cannot recognize it - if I plug a TDM400 REV E/F, everything goes well Is there anybody out there who can help me?? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]: you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card, not HiSax (well, technically, you could use HiSax too, but avoid that if possible). I prefered to use hisax because it is already included in asteriskathome (why bristuff is not included?) bristuff-0.3 is listed as experimental, should I use 0.2 (stable)? And then... I will obtain the module zaphfc, then how to configure asterisk to use it? -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
James B. MacLean wrote: Rich Adamson wrote: From: James B. MacLean [EMAIL PROTECTED] Asterisk*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 Wildcard TDM400P REV I Board 2 OK 0 0 0 ---End of Original Message- The above does indicate a problem. The Rev E/F card is known to have issues, and most of the issues revolved around unusual failures after a week or so. But there have been several other changes leading up to the Rev I card (the latest is Rev J with only minor changes since Rev I). I don't know of anyone that has attempted to mix to Rev's of the TDM card in a system, so unknown whether that might be an issue or not. I'd contact digium support and have that Rev E/F card rma'ed under warranty. (All TDM cards are still under warranty.) Thanks for the heads up. More dissappointing is that the E/F card is the newer card purchased. Where can I go to see when certain revisions were released? Surprising that the newer card just purchased (to me) is the older rev :(. Next I'll try with just one card, but that will be another day as the machine is not local. thanks again, JES Booting with only one card did _not_ work. Tried each separately. Plugged into phone lines and not plugged into phone lines. I had expected at least that my Rev I card should have worked :(. JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Internet connection
Thanks, I will try thats. El lun, 28-11-2005 a las 17:23 -0500, C F escribió: Looks like it's losing it's connection to the DNS server, make sure you don't have any names that need to be resolved to IP address in any of the config files for asterisk. Just use IP address. There are other known ways of working around this problem (which I'm sure others will mention), but for the moment this should do. On 11/28/05, José Luis Gómez [EMAIL PROTECTED] wrote: Hello. I`m using asterisk 1.0.9 and it`s working fine until I disconect the WAN interface. Then asterisk doesn`t work fine, doesn`t make any Dial() and I don`t know where is the problem. When I connect the WAN interface all start working fine. I`m also using NAT in the same server. I don`t know what asterisk is looking for on the internet. Regards. -- José Luis Gómez Qualis Information Technology Av. Rivadavia 2553, PB Of. 43 EP 0342-4565684 int 102 Bs. As. 011-51990896 www.qualis.com.ar Soporte 0810-8880022 Santa Fe - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- José Luis Gómez Qualis Information Technology Av. Rivadavia 2553, PB Of. 43 EP 0342-4565684 int 102 Bs. As. 011-51990896 www.qualis.com.ar Soporte 0810-8880022 Santa Fe - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SNOM and 1.0.9
I have successfully upgraded to 1.2, but there is no change at all. Asterisk sees the subscriptions fine: asterisk_test*CLI sip show subscriptions Peer UserCall ID ExtensionLast state Type 195.27.242.113 320 3c26700c2e6 453 Idle dialog-info+xml 195.27.242.113 320 3c26700c2bf 451 Idle dialog-info+xml 2 active SIP subscriptions asterisk_test*CLI But does not send a message when the extensions are busy, or when ringing. When the SNOM starts, it queries Asterisk, but there seems to be no subsequent SIP packets. Any more ideas? Thanks, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Internet connection
José Luis Gómez ha scritto: Thanks, I will try thats. There was an issue in the ast_sip_ouraddrfor function. When the dns is down it fails to get the right address, you can easy patch it looking to the new code Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using Parlay VoXip SIP Gateway with Asterisk ?
I've used one with a Snom SIP server system it worked quite well but not tried it with * unfortunately. Voxtream support team are excellent though I'm sure they'll help you get it working. Robert Rozman wrote: Hi, we're having quite some problems with new hardware we're testing - Parlay Voxip ISDN-SIP gateway... So we're curious if anyone is using this in connection to Asterisk and what are experiences on this HW ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with auto dialout
Im trying to get Asterisk to send out voice alerts in conjunction with Nagios. Basically what happens is depending on the type of failure Nagios has seen a file will be created with the correct contacts phone number in the file. It will also put the correct context in the file depending on what pre-recorded message needs to be played. The file is then moved to the asterisk outgoing directory to be sent The script that gets created is as follows. ### Dial out file # Channel: IAX2/eurisp/xx Callerid: xx MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: alert-1 Extension: s Priority: 1 ## And The xxS arent in the file they contain the correct number to dial in Channel and the correct ID in Callerid. The call file above corresponds with the content below which is in extensions.conf ### [alert-1] exten = s,1,DigitTimeout,5 exten = s,2,ResponseTimeout,10 exten = s,3,Answer exten = s,4,Wait(1) exten = s,5,Playback(nagios-alert1) exten = s,6,Playback(vm-goodbye) exten = s,7,Hangup Everything seems to work fine up to the point when the call is sent out. The call is sent but never waits for the person being called to answer the phone, it just rings off after 2 or 3 rings. So the person being called never hears the recorded message. Im hoping that some here is able to give some advice on this. Here is what is seen when the call gets sent. ## -- Attempting call on IAX2/eurisp/xxx for [EMAIL PROTECTED]:1 (Retry 1) -- Call accepted by 10.0.0.3 (format gsm) -- Format for call is gsm Channel IAX2/eurisp/1 was answered. == Starting IAX2/eurisp/1 at alert-1,s,1 failed so falling back to exten 's' == Starting IAX2/eurisp/1 at alert-1,s,1 still failed so falling back to context 'default' -- Executing Playback(IAX2/eurisp/1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Macro(IAX2/eurisp/1, hangupcall) in new stack -- Executing ResetCDR(IAX2/eurisp/1, w) in new stack -- Executing NoCDR(IAX2/eurisp/1, ) in new stack -- Executing Wait(IAX2/eurisp/1, 5) in new stack -- Executing Hangup(IAX2/eurisp/1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'IAX2/eurisp/1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'IAX2/eurisp/1' -- Hungup 'IAX2/eurisp/1' Nov 29 11:54:14 NOTICE[2042]: pbx_spool.c:239 attempt_thread: Call completed to IAX2/eurisp/xxx ## Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 24/11/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIALSTATUS
Hi all, I would like to run my perl agi script when the call is hungup. I did one script to calculate calling balance and duration. I made one timer Where the dialstaus is Answered But i am am in confiuse how i can stop my timer when the dialstus will be hangup. Please give me an advice to solve my problem. -- Best Regards, Code Lover Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIALSTATUS
Hi all, How i can call my perl agi script when the call is hungup. Because i am making some external Cdr calculation. -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VegaStream
Hi Is anyone using a vegastream product with asterisk? I have various numbers coming into the vegastream vega400 and was after some exmaple config for use with the asterisk server so it can perhaps reister with the vega and recieve these numbers??? Any help or pointers in the right direction would be appreciated. Thanks Scott Pinhorne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]: you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card, not HiSax (well, technically, you could use HiSax too, but avoid that if Ok, I downloaded both bristuff-0.2 and bristuff 0.3. 0.2 don't compiled. 0.3 yes, but it broke asterisk installation Asterisk now exits with this message. Ouch ... error while writing audio data: : Broken pipe -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SNOM and 1.0.9
What's the output of show hints? office-pbx*CLI sip show subscriptions Peer UserCall ID ExtensionLast state Type 192.168.2.46 700 3c26700c5f3 703 Idle dialog-info+xml 192.168.2.46 700 3c26700c557 702 Idle dialog-info+xml 192.168.2.46 700 3c26700c530 701 Idle dialog-info+xml 3 active SIP subscriptions office-pbx*CLI show hints office-pbx*CLI -= Registered Asterisk Dial Plan Hints =- 703 : SIP/703 State:IdleWatchers 1 702 : SIP/702 State:IdleWatchers 1 701 : SIP/701 State:IdleWatchers 1 700 : SIP/700 State:IdleWatchers 0 - 4 hints registered office-pbx*CLI My lights work as expected, blinking when ringing 703/702/701 and constant on when unavailable (ie busy or not registered) Kind regards, Erik Joseph Rothstein wrote: I have successfully upgraded to 1.2, but there is no change at all. Asterisk sees the subscriptions fine: asterisk_test*CLI sip show subscriptions Peer UserCall ID ExtensionLast state Type 195.27.242.113 320 3c26700c2e6 453 Idle dialog-info+xml 195.27.242.113 320 3c26700c2bf 451 Idle dialog-info+xml 2 active SIP subscriptions asterisk_test*CLI But does not send a message when the extensions are busy, or when ringing. When the SNOM starts, it queries Asterisk, but there seems to be no subsequent SIP packets. Any more ideas? Thanks, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIALSTATUS
Code Lover wrote: Hi all, How i can call my perl agi script when the call is hungup. Because i am making some external Cdr calculation. Hi M. Lover, There are two solutions for you: - You can call an AGI on hangup by using the extension 'h' : exten = h,1,DeadAGI(myagi.agi) - If you're using the Asterisk::AGI interface, you can catch the hangup in your perl program. Have a look at http://www.voip-info.org/wiki/view/Asterisk+perl+agi in the Callbacks section. (Asterisk::Manager also provides the method setcallback() and you can catch typed callback like 'Hungup' or 'DEFAULT' but I have not tried it). Regards, Benoit -- Benoit Merouze Ingenieur Developpement d'Application Reseau [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 revisions problem: Rev J not working!!
Hi Matteo, thanks for answering, your advise seemed right but no pci or motherboard driver is avalaible on ASUS site. I think we'll use another motherboard. This is another motherboard with great problems as Dell hardware. Thanks Giorgio Incantalupo Matteo Brancaleoni wrote: sure? have you tried latest drivers? could be simply a pci-id problem. matteo. Il giorno mar, 29/11/2005 alle 11.59 +0100, gincantalupo ha scritto: Hi, I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm using a K8N-E deluxe asus motherboard which gives me some problems (but I'm not sure is the motherboard causing the problem): - if I plug a TDM400 REV J, Debian cannot recognize it - if I plug a TDM400 REV E/F, everything goes well Is there anybody out there who can help me?? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with auto dialout
I'm trying to get Asterisk to send out voice alerts in conjunction with Nagios. Basically what happens is depending on the type of failure Nagios has seen a file will be created with the correct contacts phone number in the file. It will also put the correct context in the file depending on what pre- recorded message needs to be played. The file is then moved to the asterisk outgoing directory to be sent The script that gets created is as follows. ### Dial out file # Channel: IAX2/eurisp/xx Callerid: xx MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: alert-1 Extension: s Priority: 1 ## And The xx'S aren't in the file they contain the correct number to dial in Channel and the correct ID in Callerid. The call file above corresponds with the content below which is in extensions.conf ### [alert-1] exten = s,1,DigitTimeout,5 exten = s,2,ResponseTimeout,10 exten = s,3,Answer exten = s,4,Wait(1) exten = s,5,Playback(nagios-alert1) exten = s,6,Playback(vm-goodbye) exten = s,7,Hangup Everything seems to work fine up to the point when the call is sent out. The call is sent but never waits for the person being called to answer the phone, it just rings off after 2 or 3 rings. So the person being called never hears the recorded message. I'm hoping that some here is able to give some advice on this. Here is what is seen when the call gets sent. ## -- Attempting call on IAX2/eurisp/xxx for [EMAIL PROTECTED]:1 (Retry 1) -- Call accepted by 10.0.0.3 (format gsm) -- Format for call is gsm Channel IAX2/eurisp/1 was answered. == Starting IAX2/eurisp/1 at alert-1,s,1 failed so falling back to exten 's' == Starting IAX2/eurisp/1 at alert-1,s,1 still failed so falling back to context 'default' -- Executing Playback(IAX2/eurisp/1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Macro(IAX2/eurisp/1, hangupcall) in new stack -- Executing ResetCDR(IAX2/eurisp/1, w) in new stack -- Executing NoCDR(IAX2/eurisp/1, ) in new stack -- Executing Wait(IAX2/eurisp/1, 5) in new stack -- Executing Hangup(IAX2/eurisp/1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'IAX2/eurisp/1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'IAX2/eurisp/1' -- Hungup 'IAX2/eurisp/1' Nov 29 11:54:14 NOTICE[2042]: pbx_spool.c:239 attempt_thread: Call completed to IAX2/eurisp/xxx I see two problems. First the dialplan is not finding your context. The second is that when your call is made over IAX, your box is seeing it as answered and immediately playing goodbye before it is actually answered. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SNOM and 1.0.9
Hi! I still cannot get this to work on 1.0.9. exten = 451,hint,sip/451 * Try hint,SIP/451 instead of hint,sip/451. The bugtracker has an open ticket on case-sensitivity of the hint priority. * Make sure that in the advanced settings your SNOM is set to not filter packets from registrar Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Tue, November 29, 2005 13:17, Alejandro Vargas said: 2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]: you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card, not HiSax (well, technically, you could use HiSax too, but avoid that if Ok, I downloaded both bristuff-0.2 and bristuff 0.3. 0.2 don't compiled. 0.3 yes, but it broke asterisk installation Asterisk now exits with this message. Ouch ... error while writing audio data: : Broken pipe Go in to bristuff 0.3.0 directory and do ./download.sh (which downloads and patches the source) Then go to the ZapHFC subfolder and download the Florz patch there, extract it and do diff -p1 patchname Then go back to the bristuff 0.3.0 directory and do ./compile.sh This will compile and install all modules in the correct order... Works fine on my machine as we speak BTW: BRIstuff is not included by default as it breaks PRI support. Asterisk is already set up to use zap, so that is easy... Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv to start and initialize the cards... good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with auto dialout
I see two problems. First the dialplan is not finding your context. The second is that when your call is made over IAX, your box is seeing it as answered and immediately playing goodbye before it is actually answered. I think the reason it just hangs up is it falls back to the default context which is in extensions.conf: [default] include = ext-local exten = s,1,Playback(vm-goodbye) exten = s,2,Macro(hangupcall) But so is my own context I put into the file. Not sure why it can't find it Tony -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 24/11/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
Hi! on 1.0.9 the lights work. In this way: person is on the phone: light is on Person is not on the phone: light is off since 1.2 the lights will blink when the phone is running and above states work the same. Side note: Asterisk v1.2.0 comes with a new sip.conf setting: notifyringing=yes No, a typo. If the extension is ringing the led blinks, now all we need is a way to pick up that ringing channel. Could anyone tell me where the patch is that added hint support for local channels as i need to use the led for Agents (because people here don't use a fixed desk) Maybe this one? http://bugs.digium.com/view.php?id=5779 Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with auto dialout
Tony Spencer wrote: I think the reason it just hangs up is it falls back to the default context which is in extensions.conf: [default] include = ext-local exten = s,1,Playback(vm-goodbye) exten = s,2,Macro(hangupcall) I read it as if it was trying to match the context on the remote server. Hence, Attempting call on IAX2/eurisp/xxx for [EMAIL PROTECTED]:1 Isn't eurisp the remote server and alert-1 the context on that server? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
Seems like it, thnx Philipp von Klitzing wrote: Hi! on 1.0.9 the lights work. In this way: person is on the phone: light is on Person is not on the phone: light is off since 1.2 the lights will blink when the phone is running and above states work the same. Side note: Asterisk v1.2.0 comes with a new sip.conf setting: notifyringing=yes No, a typo. If the extension is ringing the led blinks, now all we need is a way to pick up that ringing channel. Could anyone tell me where the patch is that added hint support for local channels as i need to use the led for Agents (because people here don't use a fixed desk) Maybe this one? http://bugs.digium.com/view.php?id=5779 Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SNOM and 1.0.9
I changed hint using upper case SIP instead of lower case sip, and this solved my problem. Very strange indeed. Thanks to all for input. Regards, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pros and Cons of T1/E1 cards
David Waugh wrote: This means for example that the card could be used for a conferencing application with 24 users with echo cancellation/ gain control being handled by the card - and not having to be processed by the central CPU. That is correct, of course, but keep in mind that having enough CPU horsepower to do those functions on the host will cost less than $1000US more than a system that couldn't (and that's assuming your low end under $1000US box cannot do it... many of them can). This is the reason why even Intel/Dialog has moved towards 'host media processing' instead of DSP-laden boards... the DSPs are just more expensive per unit than doing the same work on the host CPU. Dedicated ASICs (like echo cancellers and conferencing engines) appear to still have a market, but using general purpose DSPs for these functions is no longer cost-effective. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/29, Francesco Peeters [EMAIL PROTECTED]: Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv to start and initialize the cards... I'll try... when somebody goes to reset the machine. I'm configuring it through ssh and it hanged when I was trying zaphfc module. The lastest problems I had where asterisk didn't start. The error was it could'nt access the device. Lastest problems I had (after reinstalling asteriskathome and removing some modules) where like ZT_SPANCONFIG failed on span 1: Invalid argument (22) in this case: [EMAIL PROTECTED] zaphfc]# ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) And trying to start asterisk I received this errors: Nov 29 14:14:25 WARNING[4240] chan_zap.c: Unable to specify channel 1: Device or resource busy Nov 29 14:14:25 ERROR[4240] chan_zap.c: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 Nov 29 14:14:25 ERROR[4240] chan_zap.c: Unable to register channel '1-2' Nov 29 14:14:25 WARNING[4240] loader.c: chan_zap.so: load_module failed, returning -1 Nov 29 14:14:25 WARNING[4240] loader.c: Loading module chan_zap.so failed! Ahd ztcfg -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk cdr mysql
hum, may be a mismatching between the asterisk source and the mysql module source. Where are you getting the sources and please explain how are you starting the compilation. Best RegardsOn 11/27/05, Abdul Lateef Khan [EMAIL PROTECTED] wrote: Hi all,Did anyone installed asterisk-addons successfull? Becuase i am gettingsome error in installation.Error:cdr_addon_mysql.c: In function `my_load_module':cdr_addon_mysql.c:292: warning: assignment makes pointer from integer without a castcc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o-lmysqlclient -lz-L/usr/lib/mysqlcc -fPIC -I../asterisk -D_GNU_SOURCE-I/usr/include/mysql -c -oapp_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4arguments, but only 3 givenapp_addon_sql_mysql.c: In function `del_identifier':app_addon_sql_mysql.c:164: `AST_LIST_REMOVE' undeclared (first use in this function)app_addon_sql_mysql.c:164: (Each undeclared identifier is reported only onceapp_addon_sql_mysql.c:164: for each function it appears in.)make: *** [app_addon_sql_mysql.o] Error 1rm app_saycountpl.o Please help me how i can load this mysql cdr module?--Best Regards,Abdul Lateef KhanComputer ProgrammerMobile No. : +974 - 5405022ICQ : 276-994-704YM! : [EMAIL PROTECTED]MSN : [EMAIL PROTECTED]Google Talk : [EMAIL PROTECTED]___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with Channels
First remember that for each change in zapata.conf you must restart asterisk, not only reload configuration. Now, could you provide a link to show us your zaptel.con and zapata.conf? when you type ztcfg -vv ? what does the output says exactly? best regardsOn 11/26/05, Scott Geist [EMAIL PROTECTED] wrote: Before using asterisk I can see that all the channels are set correctly on the digium wildcards. But when running Asterisk doing a 'zap show channels' shows them as unconfigured. There is three cards total and all are seen outside of asterisk with a total of 12 channels, but when in asterisk it sees the cards but says unconfigured. What did I miss or what did I do wrong. I tried to follow the installation guide I had perfectly. Scott ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] qozap.o error
I am trying to install the qozap driver, but when I doing: make all the shell command show error in qozap.o.What can I doing for compiling qozap.o?Thanks Yahoo! Messenger: chiamate gratuite in tutto il mondo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
zahfc mode loaded ? try lsmod to verify try ztcfg -vvv sleep 3 ztcfg -vvv 2005/11/29, Alejandro Vargas [EMAIL PROTECTED]: 2005/11/29, Francesco Peeters [EMAIL PROTECTED]: Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv to start and initialize the cards... I'll try... when somebody goes to reset the machine. I'm configuring it through ssh and it hanged when I was trying zaphfc module. The lastest problems I had where asterisk didn't start. The error was it could'nt access the device. Lastest problems I had (after reinstalling asteriskathome and removing some modules) where like ZT_SPANCONFIG failed on span 1: Invalid argument (22) in this case: [EMAIL PROTECTED] zaphfc]# ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) And trying to start asterisk I received this errors: Nov 29 14:14:25 WARNING[4240] chan_zap.c: Unable to specify channel 1: Device or resource busy Nov 29 14:14:25 ERROR[4240] chan_zap.c: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 Nov 29 14:14:25 ERROR[4240] chan_zap.c: Unable to register channel '1-2' Nov 29 14:14:25 WARNING[4240] loader.c: chan_zap.so: load_module failed, returning -1 Nov 29 14:14:25 WARNING[4240] loader.c: Loading module chan_zap.so failed! Ahd ztcfg -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problem with ADIT 600 and FXO configuration
Couple of other items to look at... the 'zap show channels' should look
something like:
pseudoinbound-bus-lin en default
1inbound-bus-dia en default
I don't see the 'Language' colume on your display below. Does your
zaptel.conf include:
loadzone = us
defaultzone=us
and your zapata.conf include:
[channels]
language=en
I'm using cvs-head and never have played with any of the stable versions,
so could be a difference in some of these commands and displays. Asterisk
v1.2 is very close to (if not identical) to cvs-head as of today, but
won't remain that way for very long.
What do you mean Yes the calls out are/were to Zap/g1/xxx?
Your outbound extensions.conf entry should look something like:
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1})
What is xx in your example? Copy/paste the exact entry that
you are trying to use.
Attach a voltmeter/test set across the outbound tip/ring and watch for
the needle either going to zero, or, reversing polarity about the same
time the dtmf digits are sent. Do you see anything other then maybe a
solid 10 to 15 volts? If you do see a dip to zero volts or a reversal
of polarity, then its highly likely the pstn line is using some sort of
signaling that you've not addressed as yet. (eg, maybe gound start, or
the line might have a form of voicemail notification on it from the
central office that was intended to blink an LED on an analog phone. I
actually used a service like that for a year or so from an Alltel
electronic central office.)
Rich
Hello,
On Tue, 2005-11-29 at 02:25, Rich Adamson wrote:
Well... I don't have an ADIT box around, so can't help on that.
Do take a close look at the channel assignment stuff, both in zaptel.conf
and zapata.conf. Are you absolutely sure the ordering of the cards
and channels are right (haven't moved any cards around or removed any)?
Your statement it wasn't until I changed the connection to span 2 that
it started allowing inbound calls to work suggests the ordering of
the channels might not be what you are expecting.
You have channels 25-27 defined in zapata.conf, but they are shown as
unused in zaptel.conf. (I did not try to match up all the other ones.)
Sorry, I had also make the requisite changes in zaptel.conf:
span=1,0,0,esf,b8zs,yellow
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
fxsks=1-8
unused=9-16
unused=17-24
fxsks=25-48
unused=49-72,73-96
fxsks=97
fxoks=98-101
fxoks=102-105
loadzone = us
defaultzone=us
Take a close look at the group= definitions below. First set to
group=1, then six lines below that its group=0. Are you calling out
with an extensions.conf entry like Zap/g1? And, are all the channels
that are included in g1 actually connected/usable? (eg, be carefull
with assumptions about what happens when a channel is included in the
group definition but the associated ADIT port isn't connected to
anything.) Instead of using Zap/g1, prove to yourself things are
configured correctly by sending calls to Zap/99 (or whichever channel
you have connected to a real line), and do that for each fxo line
that you think is wired/working.
Yes the calls out are/were to Zap/g1/xxx, changing them to
the specific Zap channels makes no difference. I just now
tried adding w to the dial stream, no effect. Discovered
that my new test-set shows DTMF digits, hooked it up and
I'm seeing only the first digit of the phone number being
sent on the outgoing line (the reason for the Call didn't
go through message). Any ideas where next to look?
Might look at 'zap show status' and 'zap show channels' to ensure
what your expecting is what is defined.
Is show status a asterisk 1.2 command?
*CLI zap show status
No such command 'zap show status' (type 'help' for help)
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudointernal default
25incoming default
26incoming default
27incoming default
97incoming default
98internal default
99internal default
100internal default
101internal default
102internal default
103internal default
104internal default
105internal default
*CLI
RED/NOP: RED generally means the T1 port is not seeing any timing
signals (eg, nothing is connected to it). NOP generally mean
Not-OPerational.
When the cable is connected to span1 the RED goes away but it
stays in NOP.
Not sure why T1 port #1 on the
[Asterisk-Users] Re: Emailed voicemail messages not being deleted
So does this problem only surface with delete=yes? I am using 1.0.9 and do not have the second comma. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Dustin Wenz [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] That appears to have done the trick...I guess I expected some sort of warning at the console if I had inadvertently malformed the parameter string. It works now though, so it's all good. Thanks for the help! - .Dustin Wenz On Nov 28, 2005, at 2:15 PM, Gonzalo Servat wrote: On 11/28/05, Dustin Wenz [EMAIL PROTECTED] wrote: According to the Asterisk wiki, adding the delete=yes option to a voicemail definition should automatically delete messages after they are emailed. This is the format that I'm using: 101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes [snip] Try: 101 = ,First Last,[EMAIL PROTECTED],,attach=yes|delete=yes (notice the extra comma after the email address) I believe the setting that goes in between the empty commas is the pager email address Hope this helps. Cheers, Gonzalo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Wrong usage of [] in the extension?
I do not know if asterisk uses standard regexp, but in regexp you would use: [(201)(202)(203)(205)(206)] This would match any of the groups () of numbers. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Matt Riddell [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] The idea is that any number inside the [] is one checked for i.e.: _123[456]78 will match: 123478 123578 123678 -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pros and Cons of T1/E1 cards
On Tue, 29 Nov 2005, Kevin P. Fleming wrote: David Waugh wrote: This means for example that the card could be used for a conferencing application with 24 users with echo cancellation/ gain control being handled by the card - and not having to be processed by the central CPU. That is correct, of course, but keep in mind that having enough CPU horsepower to do those functions on the host will cost less than $1000US more than a system that couldn't (and that's assuming your low end under $1000US box cannot do it... many of them can). If the CPU has nothing else to do... well then it is possible. But I don't think the CPU won't have any other peaks which might disturb the echo-cancel/conference processing. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 revisions problem: Rev J not working!!
I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm using a K8N-E deluxe asus motherboard which gives me some problems (but I'm not sure is the motherboard causing the problem): - if I plug a TDM400 REV J, Debian cannot recognize it - if I plug a TDM400 REV E/F, everything goes well Is there anybody out there who can help me?? The above sounds like you are trying to use an older version of zaptel/asterisk. The pci id numbers for the Rev J card were added somewhere around v1.0.9 or so (not sure exactly which version). What asterisk version are you trying to use? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Asterisk*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 Wildcard TDM400P REV I Board 2 OK 0 0 0 ---End of Original Message- The above does indicate a problem. The Rev E/F card is known to have issues, and most of the issues revolved around unusual failures after a week or so. But there have been several other changes leading up to the Rev I card (the latest is Rev J with only minor changes since Rev I). I don't know of anyone that has attempted to mix to Rev's of the TDM card in a system, so unknown whether that might be an issue or not. I'd contact digium support and have that Rev E/F card rma'ed under warranty. (All TDM cards are still under warranty.) Thanks for the heads up. More dissappointing is that the E/F card is the newer card purchased. Where can I go to see when certain revisions were released? Surprising that the newer card just purchased (to me) is the older rev :(. You can probably search the -cvs list to find it, but that might be a little time consuming. You should see the card's pic id's in dmesg and then look in the zaptel src directory for matching entries, or, simply call digium support. It sounds like you are running an older version of zaptel/asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Presence + Eyebeam + Asterisk 1.2
Ben Buxton wrote: Can't say I've actually tried IM, but Ill give it a go sometime. I think the wiki needs updating on all this...the eyebeam page is very incomplete on subscribe, im, etc. I've got online offline status and the eyeBeam will display messages you send to it with SendText while in a call. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with auto dialout
I'm a bit of newbie to Asterisk so I'm not to sure. I was just given the task to try and make this work. You could be correct but I'd have to do some further investigation and speak to the person that used to admin this server. All I want to do is call a phone number and play a audio file and hangup. Is there a way of doing this by dropping a file in the outgoing queue or even from a script/commandline.. Thanks Tony -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 29 November 2005 13:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems with auto dialout Tony Spencer wrote: I think the reason it just hangs up is it falls back to the default context which is in extensions.conf: [default] include = ext-local exten = s,1,Playback(vm-goodbye) exten = s,2,Macro(hangupcall) I read it as if it was trying to match the context on the remote server. Hence, Attempting call on IAX2/eurisp/xxx for [EMAIL PROTECTED]:1 Isn't eurisp the remote server and alert-1 the context on that server? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 24/11/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 24/11/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: Re: Re: [Asterisk-Users] Zaptel errors on Debian
Hi, I've been using module assistant first time than using make linux26 seems OK now, meaning I still have PRI and hisax but in 2nd position and wcfxo OK. risk2:/usr/src/asterisk-1.2.0# lsmod | grep zaptel zaptel228644 1 wcfxo crc_ccitt 2144 2 zaptel,hisax Halas, I still can not install asterisk, same errors 1: chan_zap.c:10906: error: dereferencing pointer to incomplete type chan_zap.c:10906: error: dereferencing pointer to incomplete type chan_zap.c:10907: error: dereferencing pointer to incomplete type chan_zap.c:10916: error: dereferencing pointer to incomplete type chan_zap.c:10917: error: dereferencing pointer to incomplete type chan_zap.c:10932: error: dereferencing pointer to incomplete type .. Any idea or is it my pci voodoo creating zombies :- ? Hi On Mon, Nov 28, 2005 at 04:35:34PM -0800, Geo wrote: It should build wcfxo. Not trying anything special. I just follow the procedure ! When I reboot I have: ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded !!! HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed or whatever Is ISDN susbsystem needed for using fxo devices using fxs signalling with Asterisk ? What's this ISDN driver doing here? A look at lspci will show you: Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface But by now you already knowthat this line represents your X100P card that hapens to have the same PCI ID as that TigerJet device. 'lspci -n' will show that the actual device has vendor ID e159 and product ID 1 . 'grep e159 /lib/modules/`uname -r`/modules.pcimap' will show that a number of zaptel modules look for devices with those vendor/product IDs but with some specific subvendor IDs and that the hisax driver tries to load them all. hotplug uses that information (extracted from the modules at depmod time) to load modules by bus IDs. Don't want it? blacklist it: echo hisax /etc/hotplug/blacklist.d/local Consider blacklisting other modules whose automatic modprobe seems unnecessary/pointless in just the same way (or $EDITOR /etc/hotplug/blacklist ) than Zapata Telephony Interface Registered on major 196 wcfxo: disagrees about version of symbol zt_receive wcfxo: Unknown symbol zt_receive wcfxo: disagrees about version of symbol zt_ec_chunk wcfxo: Unknown symbol zt_ec_chunk wcfxo: disagrees about version of symbol zt_transmit This beats me: version mipatch between zaptel and wcfxo ? One possible guess: you installed everything from one place. And then you compiled it again (without wcfxo this time) and reinstalled. Are you using m-a? .. Testing modprobe zaptel = OK zaptel driver but not wcfxo and ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Seems that wcfxo is not loaded. lsmod | grep zaptel Yet my config is OK. = Installind Asterisk make install compiling OK but errors on zap .. chan_zap.c:8935: error: dereferencing pointer to incomplete type chan_zap.c:8936: error: dereferencing pointer to incomplete type chan_zap.c:8950: error: dereferencing pointer to incomplete type .. On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote: Well, thanks, it might be great your package yet I would like to know how to adapt. I wouldn't like to rewrite Debian neither Asterisk but is somebody able to advice how you define modules in zconfig.h or whatever ? Any tip ? Geo Why would you need to define modules? The package builds wcfxo. What exactly do you try to do? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: Re: Re: Re: [Asterisk-Users] Zaptel errors on Debian
Hi, I've been using module assistant first time than using make linux26 seems OK now, meaning I still have PRI and hisax but in 2nd position and wcfxo OK. risk2:/usr/src/asterisk-1.2.0# lsmod | grep zaptel zaptel228644 1 wcfxo crc_ccitt 2144 2 zaptel,hisax Halas, I still can not install asterisk, same errors 1: chan_zap.c:10906: error: dereferencing pointer to incomplete type chan_zap.c:10906: error: dereferencing pointer to incomplete type chan_zap.c:10907: error: dereferencing pointer to incomplete type chan_zap.c:10916: error: dereferencing pointer to incomplete type chan_zap.c:10917: error: dereferencing pointer to incomplete type chan_zap.c:10932: error: dereferencing pointer to incomplete type .. Any idea or is it my pci voodoo creating zombies :- ? Hi On Mon, Nov 28, 2005 at 04:35:34PM -0800, Geo wrote: It should build wcfxo. Not trying anything special. I just follow the procedure ! When I reboot I have: ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded !!! HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed or whatever Is ISDN susbsystem needed for using fxo devices using fxs signalling with Asterisk ? What's this ISDN driver doing here? A look at lspci will show you: Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface But by now you already knowthat this line represents your X100P card that hapens to have the same PCI ID as that TigerJet device. 'lspci -n' will show that the actual device has vendor ID e159 and product ID 1 . 'grep e159 /lib/modules/`uname -r`/modules.pcimap' will show that a number of zaptel modules look for devices with those vendor/product IDs but with some specific subvendor IDs and that the hisax driver tries to load them all. hotplug uses that information (extracted from the modules at depmod time) to load modules by bus IDs. Don't want it? blacklist it: echo hisax /etc/hotplug/blacklist.d/local Consider blacklisting other modules whose automatic modprobe seems unnecessary/pointless in just the same way (or $EDITOR /etc/hotplug/blacklist ) than Zapata Telephony Interface Registered on major 196 wcfxo: disagrees about version of symbol zt_receive wcfxo: Unknown symbol zt_receive wcfxo: disagrees about version of symbol zt_ec_chunk wcfxo: Unknown symbol zt_ec_chunk wcfxo: disagrees about version of symbol zt_transmit This beats me: version mipatch between zaptel and wcfxo ? One possible guess: you installed everything from one place. And then you compiled it again (without wcfxo this time) and reinstalled. Are you using m-a? .. Testing modprobe zaptel = OK zaptel driver but not wcfxo and ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Seems that wcfxo is not loaded. lsmod | grep zaptel Yet my config is OK. = Installind Asterisk make install compiling OK but errors on zap .. chan_zap.c:8935: error: dereferencing pointer to incomplete type chan_zap.c:8936: error: dereferencing pointer to incomplete type chan_zap.c:8950: error: dereferencing pointer to incomplete type .. On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote: Well, thanks, it might be great your package yet I would like to know how to adapt. I wouldn't like to rewrite Debian neither Asterisk but is somebody able to advice how you define modules in zconfig.h or whatever ? Any tip ? Geo Why would you need to define modules? The package builds wcfxo. What exactly do you try to do? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
[Asterisk-Users] moh on optipoint400
Hi all, i'm wondering if anyone has ever managed to get moh working on Siemens OptiPoint400? if yes, can you please explain how you did it... thx. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID Block (*67)
Client wants to use a *67 feature to block caller id on next call. In the Wiki I have seen references to this being available but I haven't see any code to actually make it work. Does anyone have a quick solution for implementing this type of function? -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID Block (*67)
Assuming AMP style contexts: PRI: [from-internal] exten = *67,1,SetCallerID( ) exten = *67,2,SetCallerIDName( ) exten = *67,3,SetCallerIDNum( ) exten = *67,4,Playback(YourCustomPromptStar67IsEnabled) exten = *67,5,DISA(no-password|from-internal) POTS: [from-internal] exten = *67,1,Dial(ZAP/1/*67 ) exten = *67,2,Wait(3) exten = *67,3,SoftHangup(ZAP/1) exten = *67,4,Playback(YourCustomPromptStar67IsEnabled) exten = *67,5,DISA(no-password|from-internal) Untested, but don't see why it shouldn't work hth -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 29, 2005 8:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Caller ID Block (*67) Client wants to use a *67 feature to block caller id on next call. In the Wiki I have seen references to this being available but I haven't see any code to actually make it work. Does anyone have a quick solution for implementing this type of function? -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: Re: Re: [Asterisk-Users] Zaptel errors on Debian
On Tue, Nov 29, 2005 at 04:35:27PM -0800, Geo wrote: Hi, I've been using module assistant first time than using make linux26 seems OK now, meaning I still have PRI and hisax The module hisax is harmless. Ignore it. Black-list it (see my previous mail) if it bothers you ion the logs and start-time, but apart from that: nothing to worry about. but in 2nd position and wcfxo OK. risk2:/usr/src/asterisk-1.2.0# lsmod | grep zaptel zaptel228644 1 wcfxo crc_ccitt 2144 2 zaptel,hisax Halas, I still can not install asterisk, same errors 1: Good, so we've passed zaptel, and got to Asterisk. Seems like yo need to start providing more details here. And this seems like a good time as any to mention: echo deb http://rapid.dotsrc.org/ unstable/ /etc/apt/sources.list echo deb http://rapid.dotsrc.org/ experimental/ /etc/apt/sources.list apt-get update apt-get install asterisk Might actually even work... chan_zap.c:10906: error: dereferencing pointer to incomplete type chan_zap.c:10906: error: dereferencing pointer to incomplete type chan_zap.c:10907: error: dereferencing pointer to incomplete type chan_zap.c:10916: error: dereferencing pointer to incomplete type chan_zap.c:10917: error: dereferencing pointer to incomplete type chan_zap.c:10932: error: dereferencing pointer to incomplete type .. Any idea or is it my pci voodoo creating zombies :- ? Hmmm. what zombies? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID Block (*67)
I will install it and test it. Thanks. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Tuesday, November 29, 2005 8:15 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Caller ID Block (*67) Assuming AMP style contexts: PRI: [from-internal] exten = *67,1,SetCallerID( ) exten = *67,2,SetCallerIDName( ) exten = *67,3,SetCallerIDNum( ) exten = *67,4,Playback(YourCustomPromptStar67IsEnabled) exten = *67,5,DISA(no-password|from-internal) POTS: [from-internal] exten = *67,1,Dial(ZAP/1/*67 ) exten = *67,2,Wait(3) exten = *67,3,SoftHangup(ZAP/1) exten = *67,4,Playback(YourCustomPromptStar67IsEnabled) exten = *67,5,DISA(no-password|from-internal) Untested, but don't see why it shouldn't work hth -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 29, 2005 8:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Caller ID Block (*67) Client wants to use a *67 feature to block caller id on next call. In the Wiki I have seen references to this being available but I haven't see any code to actually make it work. Does anyone have a quick solution for implementing this type of function? -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] qozap.o error
On Tue, Nov 29, 2005 at 03:56:24PM +0100, asterisk183 wrote: I am trying to install the qozap driver, but when I doing: make all the shell command show error in qozap.o. What can I doing for compiling qozap.o? Thanks Start by giving the telepathy-chalanged among us some clues of your setup: - version of asterisk - version of bristuff - Linux version - what did you do so far? I should note that for Debian Sarge I already have them pre-built and packaged. But then again, you probably don't want to miss the fun. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Caller ID Block (*67)
Could you just use a different start number? 9 to dial out. 8 to dial out with blocked callerID. Then just preface the callerID block code for the Telco. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Colin Anderson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Assuming AMP style contexts: PRI: [from-internal] exten = *67,1,SetCallerID( ) exten = *67,2,SetCallerIDName( ) exten = *67,3,SetCallerIDNum( ) exten = *67,4,Playback(YourCustomPromptStar67IsEnabled) exten = *67,5,DISA(no-password|from-internal) POTS: [from-internal] exten = *67,1,Dial(ZAP/1/*67 ) exten = *67,2,Wait(3) exten = *67,3,SoftHangup(ZAP/1) exten = *67,4,Playback(YourCustomPromptStar67IsEnabled) exten = *67,5,DISA(no-password|from-internal) Untested, but don't see why it shouldn't work hth -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 29, 2005 8:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Caller ID Block (*67) Client wants to use a *67 feature to block caller id on next call. In the Wiki I have seen references to this being available but I haven't see any code to actually make it work. Does anyone have a quick solution for implementing this type of function? -Kerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives
Rich Adamson wrote: Thanks for the heads up. More dissappointing is that the E/F card is the newer card purchased. Where can I go to see when certain revisions were released? Surprising that the newer card just purchased (to me) is the older rev :(. You can probably search the -cvs list to find it, but that might be a little time consuming. You should see the card's pic id's in dmesg and then look in the zaptel src directory for matching entries, or, simply call digium support. It sounds like you are running an older version of zaptel/asterisk. Thanks again Rich for the info. This is all from latest CVS though. I have generated an e-mail support ticket with digium, so I am looking forward to the answer. No doubt it will be too obvious :). JES begin:vcard fn:James B MacLean n:MacLean;James B org:Education;ITS Technical Services adr:;;;Halifax;NS;;Canada email;internet:[EMAIL PROTECTED] url:http://www.ednet.ns.ca/~macleajb version:2.1 end:vcard smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring Zaptel Errors
Is there a way to monitor zaptel errors with something like Nagios? I have a TE405P and seldomly I see messages like this: Zaptel: Master changed to TE4/0/1 wct4xxp: Setting yellow alarm on span 4 wct4xxp: Clearing yellow alarm on span 4 which means that somehow the T1 went down and came back up and also means that any active call on that circuit was dropped. I'd like to be able to log/graph the frequency on these T1s failing other than by manually executing dmesg. Any ideas? Thanks, Waldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ResetCDR with CDR
Hi,
I am trying to execute the following asterisk command from one of my AGI script.
By providing 'C' flag, I exected CDR would reset.
Problem is, CDR was reset but CDR didn't grab destination number (extension)
from the Dial command.
Well my AGI script was executed after answering a call on a channel.
EXEC DIAL IAX2/{context}/{Extention}|45|CH
What I am missing?
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Re: [Asterisk-Users] qozap.o error
I risolved my problem: I have kernel source in /usr/src/linux-2.4.686 instead of /usr/src/linux, therefore the qozap.c doesn't compiling.ThanksTzafrir Cohen [EMAIL PROTECTED] ha scritto: On Tue, Nov 29, 2005 at 03:56:24PM +0100, asterisk183 wrote: I am trying to install the qozap driver, but when I doing: make all the shell command show error in qozap.o. What can I doing for compiling qozap.o? ThanksStart by giving the telepathy-chalanged among us some clues of yoursetup:- version of asterisk- version of bristuff- Linux version- what did you do so far?I should note that for Debian Sarge I already have them pre-built andpackaged. But then again, you probably don't want to miss the fun.-- Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | bestICQ# 16849755 | | friend___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office with all employee's offsite
OK, then this is easy. Instal Asterisk in the central location, along with a Sipura SPA-3000. Configure that unit to answer the incoming POTS line and act as a VOIP gateway for Asterisk. Then configure two additional SPA-3000 units, one at each employee's location. Then, configure Asterisk (I recommend [EMAIL PROTECTED] for your setup, BTW) to route the incoming call to the right extension based on time of day, auto-attendant, whatever. The SPA-3000 units at each remote site will also be able to accept the employee's incoming POTS line and pass that call through to the phone they normally use without resorting to sending it to the Asterisk server and back. (It's all in the SPA-3000 setup. Very cool indeed. Thanks Tom! Now to throw a monkey-wrench into the works... One of the employees spends a lot of time outside of his home office, and is then reachable only by cell phone. But we (for obvious reasons) don't want to hand out his cell number to everyone who wants to reach him. So, he will often forward his home phone to his cell, and forward the main office number to his home number (so when people call the office, they get his cell without realizing it). Is there any way to use the SPA-3000 at his house to re-route calls (VOIP calls, in this case) to his cell? Or would that have to be done at the office where the server is physically. I'm not clear on whether the Asterisk server can control a remote SPA-3000 in this way. I guess this could be done directly from the Asterisk server, couldn't it? It wouldn't be something that could happen automatically; it would have to be manually turned on and off. But it would also require another POTS line at the main office for the outbound call -- so I'd rather leverage the phone line at his home office to make the outgoing call to his cell phone if at all possible... One more monkey-wrench -- what if I want both of the employees to be on the phone at the same time? Two incoming POTS lines, and two SPA-3000's at the office? Or does it make more sense at that time to get a TDMxx card? This will not change, you're still looking at three lines in the scenario I outlined above. (Unless you switch to incoming VOIP, but I do *NOT* recommend that.) Nope, I don't believe in VOIP replacing POTS completely yet. Maybe in 5 years... =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | Jason Marshall, [EMAIL PROTECTED] Spots InterConnect, Inc. Calgary, AB | =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comedian Voicemail? PROBLEMS?
What's the 'format' line of the [general] section of your voicemail.conf? Martin Joseph wrote: On Nov 28, 2005, at 3:55 PM, BJ Weschke wrote: On 11/28/05, Martin Joseph [EMAIL PROTECTED] wrote: snipI am only able to get comedian voicemail (ie dialing 1234) to record or playback messages if I use the GSM codec? Is this normal and expected? If I use ulaw or alaw I get either trash noise or an immediate busy signal on attempted message playback. I am running asterisk 1.2 on OSX 10.4.3. snip This is definitely not normal or expected. Are there any errors that come up on the CLI? snip It seems to be running along smoothly until it attempts playback and then... Nov 29 02:22:35 WARNING[38]: format_wav.c:153 check_header: Not a wav file 49 Nov 29 02:22:35 WARNING[38]: file.c:432 ast_filehelper: Unable to open file on /var/spool/asterisk/voicemail/default/1234/Old/msg.wav Nov 29 02:22:35 WARNING[38]: file.c:820 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/1234/Old/msg (format alaw): No such file or directory == Spawn extension (autocontext, 8500, 1) exited non-zero on 'IAX2/2001-2' -- Hungup 'IAX2/2001-2' I do appreciate the attention and hopefully helpful suggestions? Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 revisions problem: Rev J not working!!
Actually, Matteo meant zaptel drivers, not motherboard or chipset drivers from ASUS :) Mojo gincantalupo wrote: Hi Matteo, thanks for answering, your advise seemed right but no pci or motherboard driver is avalaible on ASUS site. I think we'll use another motherboard. This is another motherboard with great problems as Dell hardware. Thanks Giorgio Incantalupo Matteo Brancaleoni wrote: sure? have you tried latest drivers? could be simply a pci-id problem. matteo. Il giorno mar, 29/11/2005 alle 11.59 +0100, gincantalupo ha scritto: Hi, I'm trying to setup an asterisk-based PBX on a Debian Sarge distro. I'm using a K8N-E deluxe asus motherboard which gives me some problems (but I'm not sure is the motherboard causing the problem): - if I plug a TDM400 REV J, Debian cannot recognize it - if I plug a TDM400 REV E/F, everything goes well Is there anybody out there who can help me?? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Caller ID Block (*67)
Actually, why not:
exten = *67XXX,1, {etc}
-Original Message-
From: Steven [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 29, 2005 9:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Caller ID Block (*67)
Could you just use a different start number?
9 to dial out. 8 to dial out with blocked callerID.
Then just preface the callerID block code for the Telco.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Colin Anderson [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
.
Assuming AMP style contexts:
PRI:
[from-internal]
exten = *67,1,SetCallerID( )
exten = *67,2,SetCallerIDName( )
exten = *67,3,SetCallerIDNum( )
exten = *67,4,Playback(YourCustomPromptStar67IsEnabled)
exten = *67,5,DISA(no-password|from-internal)
POTS:
[from-internal]
exten = *67,1,Dial(ZAP/1/*67 )
exten = *67,2,Wait(3)
exten = *67,3,SoftHangup(ZAP/1)
exten = *67,4,Playback(YourCustomPromptStar67IsEnabled)
exten = *67,5,DISA(no-password|from-internal)
Untested, but don't see why it shouldn't work hth
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 29, 2005 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Caller ID Block (*67)
Client wants to use a *67 feature to block caller id on next call. In the
Wiki I have seen references to this being available but I haven't see any
code to actually make it work. Does anyone have a quick solution for
implementing this type of function?
-Kerry
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Re: [Asterisk-Users] Load spikes with 1.0.10
Are your interrupts getting hogged by anything else? I'd recommend
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting if
you haven't already read it. Have you tried booting with noapic kernel
option? You may then have to shuffle cards around to make your sangoma
not share any interrupts shrug hth :)
moj
Gavin Hamill wrote:
Hi, I have a trivial setup on a 2.4GHz Xeon Dell PE 1750 SCSI machine
dealing with 4 ports of E1 in an 'inline PBX' arrangement.
My extensions.conf is simply:
[general]
static=yes
writeprotect=yes
[frompstn]
exten = _31.,1,Dial(Zap/g2/${EXTEN})
exten = _31.,2,Congestion
[fromaxxess]
exten = _13.,1,Dial(SIP/${EXTEN},,h)
exten = _13.,2,Congestion
exten = _31.,1,Dial(Zap/g2/${EXTEN})
exten = _31.,2,Congestion
include = outbound
[outbound]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
We have a full 30-channel PRI and a 4-channel partial PRI and are
experiencing load spikes that I can't find the source of.
The machine Debian sarge on the default 2.6.8-2-686 kernel, and no other
daemons are running than sshd.
The machine is doing no IP work - purely TDM, yet on a Xeon 2.4GHz
machine, the load average is sitting at 0.6 with 40 active Zap channels
(i.e. 20 live calls) and will randomly jump to 2 (with call quality
starting to stutter)
A few seconds of vmstat:
procs ---memory-- ---swap-- -io --system--
cpu
r b swpd free buff cache si sobibo incs us sy
id wa
7 0 0 223560 1276 22304000 310 8394 1 2
97 0
0 0 0 223552 1284 22304000 016 5128 3461 1 0
98 1
0 0 0 223552 1284 22304000 0 0 5094 3319 10 9
81 0
0 0 0 223552 1284 22304000 016 5130 2955 1 10
89 0
0 0 0 223552 1292 22304000 060 5121 2918 0 1
97 2
0 0 0 223552 1292 22304000 0 0 5031 2936 1 0
99 0
Does this sound about normal for what is just shuffling data between
ports of the Sangoma A104? I want to record the call data with the
'Monitor' application but this just causes the load to increase even
more (even though 'hdparm' shows 70MB/sec disk transfer with low
user+system CPU usage)
/proc/interrupts is
CPU0
0: 423253622IO-APIC-edge timer
1:175IO-APIC-edge i8042
9: 0 IO-APIC-level acpi
11: 0 IO-APIC-level ohci_hcd
12: 58IO-APIC-edge i8042
15: 13IO-APIC-edge ide1
177: 50 IO-APIC-level ioc0
185: 29 IO-APIC-level ioc1
193: 1311243931 IO-APIC-level wanpipe1, wanpipe2, wanpipe3, wanpipe4
201: 13289965 IO-APIC-level eth0
217:5420038 IO-APIC-level eth2
NMI: 0
LOC: 423311408
ERR: 0
MIS: 0
Help! :)
Cheers,
Gavin.
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan Company, LLC
(907) 747- x112
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[Asterisk-Users] Voicepulse not accepting new customers. (FCC E911)
Hello All, It seems that voicepulse is not taking any new orders on the standard service plans (though vp connect seems unaffected) due to the fcc rulings. We'll see what happens, anyone having similar problems with other services as of today? Greg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM and 1.0.9
On 09:46, Tue 29 Nov 05, Erik wrote: Leif Neland wrote: On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote: From memory (at a previous installation) you will need a newer version of Asterisk than 1.09 for the lights to work. on 1.0.9 the lights work. In this way: person is on the phone: light is on Person is not on the phone: light is off since 1.2 the lights will blink when the phone is running and above states work the same. Running? Is that a 3. state? No, a typo. If the extension is ringing the led blinks, now all we need is a way to pick up that ringing channel. Could anyone tell me where the patch is that added hint support for local channels as i need to use the led for Agents (because people here don't use a fixed desk) Thnx Erik. It was indeed a typo. I've read several ways to do the pickup, cant remember where right now. I know there's an entry on the mantis site. Also the bristuffed package has some notes about it. As stated in the BRIstuffed CHANGES: - SNOM call pick up with blinking LEDs (extension hints): - configure a SNOM function key as destination, for example 100 - set up an extension hint: exten = 100,hint,SIP/somePhone - and an extension: exten = 100,1,Dial(SIP/somePhone) exten = 100,2,Hangup - forget about callgroups and pickupgroups! - set up a pickup exten : exten = *8100,1,PickUpChan(SIP/somePhone) - if SIP/somePhone is idle you press the destination button to call extension 100 - if SIP/somePhone is ringing you press the button to do a pickup by calling extension *8100 Good luck -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VegaStream
On Tue, 29 Nov 2005 06:14:54 +, scott wrote: Is anyone using a vegastream product with asterisk? I have various numbers coming into the vegastream vega400 and was after some exmaple config for use with the asterisk server so it can perhaps reister with the vega and recieve these numbers??? Any help or pointers in the right direction would be appreciated. I only followed the Step by step configuration on the cd. The following file: Initial config - R7 Vega 400 E1_T1 (SIP)_03.pdf And then in added the ip for my * as Default Proxy Host Name/IP in the SIP settings. I don't use the registration at all. Then added this in sip.conf (actually AMP): [vega] type=user dtmfmode=inband disallow=all context=from-vega allow=alaw [vega-gw] type=peer host=192.168.102.37 dtmfmode=inband disallow=all context=from-vega allow=alaw I had to remove the VAD on the vega and change the dtmf settings as well. But we are since then very happy with the vega400. /Niklas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM Phones MWI, Hold Retrieve buttons not working with Asterisk v1.2
On 18:26, Mon 28 Nov 05, Sascha Deri wrote:
I made an error in what I previously wrote. What actually works in v1.2 is:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
Which is what Michael originally wrote. My bad!
:) To err is human :)
I know for sure it had to work since I copied it from my
working config.
Sascha wrote:
Thanks Michael - you got me on the right path. What you gave me didn't
work, but I figured out that the following does (on version 1.2):
exten = default,1,VoiceMailMain(${CALLERIDNUM})
(BTW, exten = Unknown,1,VoiceMailMain(${CALLERIDNUM}) used to work
for us in Asterisk 1.0.9 but obviously no longer does)
Like I said those are the defaults.
If memory serves right there's a setting for sip.conf to
specify the user that sends the MWI stuff to the phone.
Pressing the Mailbox button calls that user in this way:
[EMAIL PROTECTED]
So you have to setup an extension in the phones context that
matches that username.
The defaults were Unknown for 1.0.x and asterisk for 1.2
This is just to be complete ;)
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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Re: [Asterisk-Users] Newbie question on 1.2 extension configs
On 00:24, Tue 29 Nov 05, bram kortleven wrote: Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Get the source of asterisk and type: make samples That will create a set of default config files. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Tue, November 29, 2005 16:04, Giovanni Miano said: zahfc mode loaded ? try lsmod to verify try ztcfg -vvv sleep 3 ztcfg -vvv Also helpful is cat /proc/zaptel/* This'll tell you whether zaptel is loaded, whether the channels have been defined, and what their status is... Here's an example of mine: Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) Span 2: ZTHFC2 HFC-S PCI A ISDN card 1 [TE] layer 1 ACTIVATED (F7) AMI/CCS 4 ZTHFC2/0/1 Clear (In use) 5 ZTHFC2/0/2 Clear (In use) 6 ZTHFC2/0/3 HDLCFCS (In use) HTH -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse not accepting new customers. (FCC E911)
On Tue, November 29, 2005 18:36, [EMAIL PROTECTED] said: Hello All, It seems that voicepulse is not taking any new orders on the standard service plans (though vp connect seems unaffected) due to the fcc rulings. We'll see what happens, anyone having similar problems with other services as of today? Greg WHat fcc rulings? What did I miss? :-o -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digitmap problems
But, star at least works. I've got *xxT in my digitmap and it caught *69. In fact, my 1.5 admin guide refers to Section 2.1.5 of RFC 3435, the MGCP rfc, which does allow the * to be used Moj Rich Adamson wrote: I'm trying to implement some of the star services such as *61 for weather or *71 for wakeup call, etc. I think I have asterisk setup properly because I can get them to work fine using normal extension numbers. However, if I try to use the 'star' numbers, my Polycom IP500 never sends the digits to asterisk, at least I never see Asterisk try to do anything in the logs. I believe the phone is giving me a fast busy signal because it can not find a match in the digitmap. I've tried digitmaps like: *6x|*7x|2xxx|[2-9]x What am I missing??? The Admin Guide? I searched through the v1.5 guide, and it implies the digitmap uses numbers only (no * or #). But, it doesn't actually discuss it either. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cause 17 - User busy ?
Since upgrading to 1.2 I'm seeing the following iin my /var/log/asterisk/messages: Nov 29 11:50:20 NOTICE[13094] app_dial.c: Unable to create channel of type 'Zap' (cause 17 - User busy) Nov 29 12:02:06 WARNING[12977] chan_iax2.c: chan_iax2: ast_sched_runq ran 249 scheduled tasks all at once These may be the cause of my random disconnects of IAX calls. Can anyone provide a clue? __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_manager.conf
On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote: What is the purpose of cdr_manager.conf? cdr_manager.conf allows you to configure asterisk to send call detail records (cdr) via the Manager API. How I can configure it? to enable CDR via Manager API a cdr_manager.conf looks like this: ; ; Asterisk Call Management CDR ; [general] enabled = yes =Stefan signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small office with all employee's offsite
Jason Marshall wrote: OK, then this is easy. Instal Asterisk in the central location, along with a Sipura SPA-3000. Configure that unit to answer the incoming POTS line and act as a VOIP gateway for Asterisk. Then configure two additional SPA-3000 units, one at each employee's location. Then, configure Asterisk (I recommend [EMAIL PROTECTED] for your setup, BTW) to route the incoming call to the right extension based on time of day, auto-attendant, whatever. The SPA-3000 units at each remote site will also be able to accept the employee's incoming POTS line and pass that call through to the phone they normally use without resorting to sending it to the Asterisk server and back. (It's all in the SPA-3000 setup. Very cool indeed. Thanks Tom! Now to throw a monkey-wrench into the works... One of the employees spends a lot of time outside of his home office, and is then reachable only by cell phone. But we (for obvious reasons) don't want to hand out his cell number to everyone who wants to reach him. So, he will often forward his home phone to his cell, and forward the main office number to his home number (so when people call the office, they get his cell without realizing it). We do this all the time. We just moved and have three people working from their homes. The boss's extension rings here locally on a spare phone and rings his IAX2 phone at home. He also forwards his extension to his cellphone when he is out using *72 on the Asterisk box. One employee is working from out of state and his extension calls his cellphone. When someone dials his DID number it dials back out to his cell phone and no one knows any different. When we dial his three digit extension here it goes to his cell phone. The last person has an IAX client running on his laptop and takes calls from there. When someone calls in and presses '2' for support it rings a guy out in production and the other person working from home. I have my extension set to ring my Grandstream phone and my cell phone at the same time and I can take the calls from anywhere. I can even transfer a call back to another extension from my cellphone if they need someone else. Asterisk does all the call forwarding and phone routing. - James Is there any way to use the SPA-3000 at his house to re-route calls (VOIP calls, in this case) to his cell? Or would that have to be done at the office where the server is physically. I'm not clear on whether the Asterisk server can control a remote SPA-3000 in this way. As long as Asterisk has a way to re-dial out a phone line or voip provider, it can route an extension anywhere and the caller will not know it. I guess this could be done directly from the Asterisk server, couldn't it? It wouldn't be something that could happen automatically; it would have to be manually turned on and off. But it would also require another POTS line at the main office for the outbound call -- so I'd rather leverage the phone line at his home office to make the outgoing call to his cell phone if at all possible... One more monkey-wrench -- what if I want both of the employees to be on the phone at the same time? Two incoming POTS lines, and two SPA-3000's at the office? Or does it make more sense at that time to get a TDMxx card? This will not change, you're still looking at three lines in the scenario I outlined above. (Unless you switch to incoming VOIP, but I do *NOT* recommend that.) Nope, I don't believe in VOIP replacing POTS completely yet. Maybe in 5 years... =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | Jason Marshall, [EMAIL PROTECTED] Spots InterConnect, Inc. Calgary, AB | =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question on 1.2 extension configs
Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with auto dialout
Channel: Local/[EMAIL PROTECTED]Callerid: 01612370660MaxRetries: 5RetryTime: 300WaitTime: 45Context: serverdownExtension: sPriority: 1On 29 Nov 2005, at 15:39, Tony Spencer wrote:I'm a bit of newbie to Asterisk so I'm not to sure.I was just given the task to try and make this work.You could be correct but I'd have to do some further investigation and speakto the person that used to admin this server.All I want to do is call a phone number and play a audio file and hangup.Is there a way of doing this by dropping a file in the outgoing queue oreven from a script/commandline..ThanksTonyI have a simple system like this, the call file looks like:Channel: Local/[EMAIL PROTECTED]Callerid: 01612370660MaxRetries: 5RetryTime: 300WaitTime: 45Context: serverdownExtension: sPriority: 1SetVar: SITENAME=importantCustomerNameAnd the following in extensions.conf:[serverdown]exten = s,1,Answerexten = s,2,Wait(1)exten = s,3,Playback(serverdown/${SITENAME})exten = s,4,Wait(10)exten = s,5,Playback(serverdown/${SITENAME})exten = s,6,HangupI have a file pre-recorded with a customer specific message in serverdown/importantCustomerName.gsmThe trick with Local/[EMAIL PROTECTED] is to distribute the call to multiple users:[default]exten = 60,1,Dial(Sip/billSip/benSip/flowerSip/potSip/weed,30)Good luck,Tim. http://www.westhawk.co.uk/ ___
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[Asterisk-Users] iaxmodem
Hi everybody: Is the right behavior of the IAXmodem to display Registration completed successfully and remote hangup many times? Regards Miguel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queuelog
Entries in the queue_log file do not match what the documents say. The COMPLETECALLER and COMPLETEAGENT events do not have the 3rd agrument of origposition. I'm using Asterisk 1.0.9 currently(will be upgrading shortly). I've checked and this should be done by the old stable version we are running. We are using callback agents. Here is an example log entry: 1133290480|1133290425.5|da_queue|Agent/1|COMPLETECALLER|35|20 Here is roughly what it should be: 1133290480|1133290425.5|da_queue|Agent/1|COMPLETECALLER|35|20|1 Any reason it doing what the documents say? --johann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
Alejandro Vargas schrieb: 2005/11/29, Tomasz Chmielewski [EMAIL PROTECTED]: you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card, not HiSax (well, technically, you could use HiSax too, but avoid that if possible). I prefered to use hisax because it is already included in asteriskathome (why bristuff is not included?) you can use hisax module, so isdn4linux, but it's not very well supported by asterisk. bristuff-0.3 is listed as experimental, should I use 0.2 (stable)? use 0.3 with asterisk 1.2, 0.2 version won't work. And then... I will obtain the module zaphfc, then how to configure asterisk to use it? normally, as a zapata interface :)) although it may seem as magic, it's not that hard; if you configure zaphfc, ask here at the mailing list, or me directly, as I use it with [EMAIL PROTECTED] 2.0 -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXmodem fax polling
Adam Goryachev wrote: Don't assume that we read this list every 5 secs I haven't read the mailing list since last week You're right, thanks for your reply. In any case, you have two options: 1) Do it with meetme like you do now... Lee Howard, the author of IAXmodem agrees with me that meetme adds a layer to the call that may be bad for faxing reliability. 2) Just transfer the call to iaxmodem eg: exten = s,1,GenerateFax exten = s,2,txfax(somefax) convert to: exten = s,1,TellHylafaxWhatFaxToSend exten = s,2,Dial(IAX2/iaxmodem) Then, hylafax should answer and send the requested fax. Yes, and Hylafax will answer, *but* it will be waiting for an incoming fax, it will not try to send the fax prepared. That's why app_bridge would be needed. Steve Underwood also informed me about chan_fax (http://www.sofaswitch.org/chan_fax/), I'll have a look. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VegaStream
Hi Niklas Thanks for this information I will be sure to follow it. Many Thanks Scott Pinhorne Niklas Larsson wrote: On Tue, 29 Nov 2005 06:14:54 +, scott wrote: Is anyone using a vegastream product with asterisk? I have various numbers coming into the vegastream vega400 and was after some exmaple config for use with the asterisk server so it can perhaps reister with the vega and recieve these numbers??? Any help or pointers in the right direction would be appreciated. I only followed the Step by step configuration on the cd. The following file: Initial config - R7 Vega 400 E1_T1 (SIP)_03.pdf And then in added the ip for my * as Default Proxy Host Name/IP in the SIP settings. I don't use the registration at all. Then added this in sip.conf (actually AMP): [vega] type=user dtmfmode=inband disallow=all context=from-vega allow=alaw [vega-gw] type=peer host=192.168.102.37 dtmfmode=inband disallow=all context=from-vega allow=alaw I had to remove the VAD on the vega and change the dtmf settings as well. But we are since then very happy with the vega400. /Niklas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: IAX Call Pickup
Anyone know if this can be made to work? I've only been able to get SIP-SIP call pickup to work. Steve --- as far as I know, no. Il lun, 2004-07-05 alle 18:56, Adolfo R. Brandes ha scritto: I've looked in the obvious places but haven't found a definitive answer to the following: can an IAX extension (an Iaxy phone, for instance) do call pickup via *8? Adolfo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
Miguel Soto wrote: Is the right behavior of the IAXmodem to display Registration completed successfully and remote hangup many times? You'd have to show me an example for me to say for certain, but my guess is that if it looks wrong to you then it probably is wrong. This output should look like your Asterisk CLI output mostly. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office with all employee's offsite
I am using this dialplan with DID's to great effect, I have 130 guys doing
exactly what was discussed here. After 12 seconds ringing their SIP or IAX
client, the dialplan calls the cell automatically, during working hours. If
they don't pick up after 18 seconds, voicemail. After hours, both phones
are dialed concurrently. Also, fax detection is automatic so DID is desk
phone, cell phone, and fax.
Using it this way completely obviates the need for call forwarding; I find
CF more of a hassle than it's worth because people are dumb and forget that
calls are CF'd then complain that their phone isn't working.
Note voicemail box is the same as last 4 digits of DID. This simplifies
enduser training.
Permission given to steal this dialplan logic outright if you can put up
with my sloppy code.
[from-pstn]
;8259 is a single DID for example purposes. All this does is set variables,
then dumps the caller to a dialing context
;TODO: Make variable setting dynamically loaded from a database
exten = 8259,1,SetVar([EMAIL PROTECTED]); email
address to send faxes to
exten = 8259,2,SetVar(PRIMARYDIALSTRING=IAX2/landmark:[EMAIL PROTECTED])
;desk phone
exten = 8259,3,SetVar(SECONDARYDIALSTRING=ZAP/g0/9024985) ;cell phone
number
exten = 8259,4,SetVar(TERTIARYDIALSTRING=) ;3rd number line a home number
exten = 8259,5,SetVar(CALLRECIPIENT=Karen Kelly) ;who the person is
exten = 8259,6,SetVar(WORKSCHEDULE=SHOWHOMEHOURS) ;what their working
schedule is
exten = 8259,7,SetVar(BUILDING=BUILDING1) ;building that they report to
exten = 8259,8,SetVar(IVRVM=vm) ;after dialplan is exhausted, send them to
voicemail or to another context?
exten = 8259,9,SetVar(MAILBOX=8259) ;Mailbox number
exten = 8259,10,Goto(dial-internal,s,1)
[dial-internal]
exten = _s,1,Answer()
exten = _s,2,Wait(2);Wait 2 seconds for a fax CNG tone
exten = _s,3,Gotoif($[${WORKSCHEDULE} = BUSINESSHOURS ]?bushours,1)
exten = _s,4,Gotoif($[${WORKSCHEDULE} = SHOWHOMEHOURS
]?showhomehours,1)
exten = _s,5,Gotoif($[${WORKSCHEDULE} = SHOWHOMEHOURSSHORT
]?showhomehoursshort,1)
exten = _s,6,Goto(bushours,1);If there is no schedule set, assume Business
Hours
exten = bushours,1,Gotoiftime(*|sat|*?dialsecondary,1)
exten = bushours,2,Gotoiftime(*|sun|*?dialsecondary,1)
exten = bushours,3,Gotoiftime(8:00-17:00|mon-fri|*|*?dialprimary,1)
exten = bushours,4,Goto(dialprimary,1) ;If there's a time in this range
that doesn't fit the above, dial the Primary number anyway
exten = showhomehours,1,Gotoiftime(*|fri|*?dialsecondary,1)
exten = showhomehours,2,Gotoiftime(15:00-20:00|*|*?dialprimary,1)
exten = showhomehours,3,Gotoiftime(12:00-18:00|sat-sun|*?dialprimary,1)
exten = showhomehours,4,Goto(dialsecondary,1) ;If there's a time in this
range that doesn't fit the above, dial the Primary number anyway
exten = showhomehoursshort,1,Gotoiftime(*|fri|*?dialsecondary,1)
exten = showhomehoursshort,2,Gotoiftime(14:00-20:00|*|*?dialprimary,1)
exten =
showhomehoursshort,3,Gotoiftime(11:00-18:00|sat-sun|*?dialprimary,1)
exten = showhomehoursshort,4,Goto(dialsecondary,1) ;If there's a time in
this range that doesn't fit the above, dial the Primary number anyway
exten = dialprimary,1,SetCallerID(${CALLERIDNUM})
exten = dialprimary,2,Gotoif($[${PRIMARYDIALSTRING}foo != foo ]?3:5)
;Check for a NULL Primary Dialstring if it is null go to secondary
exten = dialprimary,3,ChanIsAvail(${PRIMARYDIALSTRING}) ; check if the
dialstring's channel is available if not go to secondary number
exten = dialprimary,4,Dial(${PRIMARYDIALSTRING},12,T)
exten = dialprimary,5,Goto(dialsecondary,1) ;If user does not pick up in 12
seconds dial his cell (secondary number)
exten = dialprimary,104,Goto(dialsecondary,1)
exten = dialsecondary,1,SetCallerID(${CALLERIDNUM})
exten = dialsecondary,2,Gotoif($[${SECONDARYDIALSTRING}foo != foo
]?3:5) ;Check for a NULL Secondary Dialstring if it is null go to tertiary
exten = dialsecondary,3,ChanIsAvail(${SECONDARYDIALSTRING}); check if the
dialstring's channel is available if not go to tertiary number
exten =
dialsecondary,4,Dial(${SECONDARYDIALSTRING}${PRIMARYDIALSTRING},18,T)
exten = dialsecondary,5,Goto(dialtertiary,1) ;If user does not pick up in
18 seconds dial his tertiary number, or voicemail
exten = dialsecondary,104,Goto(dialtertiary,1)
;Tertiary dialing not done yet, instead user is just sent to voicemail
exten = dialtertiary,1,Goto(ivr-vm,1)
exten = dialtertiary,102,Goto(ivr-vm,1)
;We can also modify the IVRVM variable to send the caller to an IVR if IVRVM
is not set to the string vm
exten = ivr-vm,1,Gotoif($[${IVRVM} = vm ]?2:3)
exten = ivr-vm,2,Voicemail([EMAIL PROTECTED])
exten = ivr-vm,3,Goto(${IVRVM},s,1)
;Inbound faxes are indicated to the user by momentarily dialing their
extension with Caller ID like this: Fax: 4035551212
;In actual use, the Primary dialstring which is typically SIP or IAX works
perfect every time
;but Secondary numbers like cell phones, the dialstring timeout is way, way
too short. Oh well.
exten =
RE: [Asterisk-Users] IAXmodem fax polling
Steve Underwood also informed me about chan_fax (http://www.sofaswitch.org/chan_fax/), I'll have a look. This looks awesome please report back to the list on this if you get it working correctly. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queuelog
Hi Johann, we engineered QueueMetrics out of the queues of * version 0.7, but never found that origposition argument. And it's not present in our current 1.2. Where did you find it? Yours l. In data Tue, 29 Nov 2005 19:57:47 +0100, Johann [EMAIL PROTECTED] ha scritto: Entries in the queue_log file do not match what the documents say. The COMPLETECALLER and COMPLETEAGENT events do not have the 3rd agrument of origposition. I'm using Asterisk 1.0.9 currently(will be upgrading shortly). I've checked and this should be done by the old stable version we are running. We are using callback agents. Here is an example log entry: 1133290480|1133290425.5|da_queue|Agent/1|COMPLETECALLER|35|20 Here is roughly what it should be: 1133290480|1133290425.5|da_queue|Agent/1|COMPLETECALLER|35|20|1 Any reason it doing what the documents say? --johann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Assum est, versa et manduca. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office with all employee's offsite
Thanks Colin, this is a fantastic list! All I need to do now is get my
butt in gear and set up the box(es)!
I am using this dialplan with DID's to great effect, I have 130 guys doing
exactly what was discussed here. After 12 seconds ringing their SIP or IAX
client, the dialplan calls the cell automatically, during working hours. If
they don't pick up after 18 seconds, voicemail. After hours, both phones
are dialed concurrently. Also, fax detection is automatic so DID is desk
phone, cell phone, and fax.
Using it this way completely obviates the need for call forwarding; I find
CF more of a hassle than it's worth because people are dumb and forget that
calls are CF'd then complain that their phone isn't working.
Note voicemail box is the same as last 4 digits of DID. This simplifies
enduser training.
Permission given to steal this dialplan logic outright if you can put up
with my sloppy code.
[from-pstn]
;8259 is a single DID for example purposes. All this does is set variables,
then dumps the caller to a dialing context
;TODO: Make variable setting dynamically loaded from a database
exten = 8259,1,SetVar([EMAIL PROTECTED]); email
address to send faxes to
exten = 8259,2,SetVar(PRIMARYDIALSTRING=IAX2/landmark:[EMAIL PROTECTED])
;desk phone
exten = 8259,3,SetVar(SECONDARYDIALSTRING=ZAP/g0/9024985) ;cell phone
number
exten = 8259,4,SetVar(TERTIARYDIALSTRING=) ;3rd number line a home number
exten = 8259,5,SetVar(CALLRECIPIENT=Karen Kelly) ;who the person is
exten = 8259,6,SetVar(WORKSCHEDULE=SHOWHOMEHOURS) ;what their working
schedule is
exten = 8259,7,SetVar(BUILDING=BUILDING1) ;building that they report to
exten = 8259,8,SetVar(IVRVM=vm) ;after dialplan is exhausted, send them to
voicemail or to another context?
exten = 8259,9,SetVar(MAILBOX=8259) ;Mailbox number
exten = 8259,10,Goto(dial-internal,s,1)
[dial-internal]
exten = _s,1,Answer()
exten = _s,2,Wait(2);Wait 2 seconds for a fax CNG tone
exten = _s,3,Gotoif($[${WORKSCHEDULE} = BUSINESSHOURS ]?bushours,1)
exten = _s,4,Gotoif($[${WORKSCHEDULE} = SHOWHOMEHOURS
]?showhomehours,1)
exten = _s,5,Gotoif($[${WORKSCHEDULE} = SHOWHOMEHOURSSHORT
]?showhomehoursshort,1)
exten = _s,6,Goto(bushours,1);If there is no schedule set, assume Business
Hours
exten = bushours,1,Gotoiftime(*|sat|*?dialsecondary,1)
exten = bushours,2,Gotoiftime(*|sun|*?dialsecondary,1)
exten = bushours,3,Gotoiftime(8:00-17:00|mon-fri|*|*?dialprimary,1)
exten = bushours,4,Goto(dialprimary,1) ;If there's a time in this range
that doesn't fit the above, dial the Primary number anyway
exten = showhomehours,1,Gotoiftime(*|fri|*?dialsecondary,1)
exten = showhomehours,2,Gotoiftime(15:00-20:00|*|*?dialprimary,1)
exten = showhomehours,3,Gotoiftime(12:00-18:00|sat-sun|*?dialprimary,1)
exten = showhomehours,4,Goto(dialsecondary,1) ;If there's a time in this
range that doesn't fit the above, dial the Primary number anyway
exten = showhomehoursshort,1,Gotoiftime(*|fri|*?dialsecondary,1)
exten = showhomehoursshort,2,Gotoiftime(14:00-20:00|*|*?dialprimary,1)
exten =
showhomehoursshort,3,Gotoiftime(11:00-18:00|sat-sun|*?dialprimary,1)
exten = showhomehoursshort,4,Goto(dialsecondary,1) ;If there's a time in
this range that doesn't fit the above, dial the Primary number anyway
exten = dialprimary,1,SetCallerID(${CALLERIDNUM})
exten = dialprimary,2,Gotoif($[${PRIMARYDIALSTRING}foo != foo ]?3:5)
;Check for a NULL Primary Dialstring if it is null go to secondary
exten = dialprimary,3,ChanIsAvail(${PRIMARYDIALSTRING}) ; check if the
dialstring's channel is available if not go to secondary number
exten = dialprimary,4,Dial(${PRIMARYDIALSTRING},12,T)
exten = dialprimary,5,Goto(dialsecondary,1) ;If user does not pick up in 12
seconds dial his cell (secondary number)
exten = dialprimary,104,Goto(dialsecondary,1)
exten = dialsecondary,1,SetCallerID(${CALLERIDNUM})
exten = dialsecondary,2,Gotoif($[${SECONDARYDIALSTRING}foo != foo
]?3:5) ;Check for a NULL Secondary Dialstring if it is null go to tertiary
exten = dialsecondary,3,ChanIsAvail(${SECONDARYDIALSTRING}); check if the
dialstring's channel is available if not go to tertiary number
exten =
dialsecondary,4,Dial(${SECONDARYDIALSTRING}${PRIMARYDIALSTRING},18,T)
exten = dialsecondary,5,Goto(dialtertiary,1) ;If user does not pick up in
18 seconds dial his tertiary number, or voicemail
exten = dialsecondary,104,Goto(dialtertiary,1)
;Tertiary dialing not done yet, instead user is just sent to voicemail
exten = dialtertiary,1,Goto(ivr-vm,1)
exten = dialtertiary,102,Goto(ivr-vm,1)
;We can also modify the IVRVM variable to send the caller to an IVR if IVRVM
is not set to the string vm
exten = ivr-vm,1,Gotoif($[${IVRVM} = vm ]?2:3)
exten = ivr-vm,2,Voicemail([EMAIL PROTECTED])
exten = ivr-vm,3,Goto(${IVRVM},s,1)
;Inbound faxes are indicated to the user by momentarily dialing their
extension with Caller ID like this: Fax: 4035551212
;In actual use, the Primary dialstring which is typically SIP or IAX works
perfect every time
;but Secondary
Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit
Yes with version 1.2. I have tried already with call-limit and the same. On 11/28/05, Kevin Hanson [EMAIL PROTECTED] wrote: Alvaro Parres wrote: Hi list...I have been testing the hint extension. And i detect that when i have in the sip.fg of the extension the incominiglimit=X (any number) the hint doesn't work all the time show the extesion as idle.If this is a bug or not ?? Thanks.___What version of Asterisk? 1.2 deprecated incominglimit in favor ofcall-limit.Cheers,Kevin___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question on Monitoring and Transferring...
Hello All, I am using * 1.2, BRIstuff 0.3 PRE1, Dual HFC-PCI, 1x TE, 1x NT I am using DECT phones on a Siemens ISDN phone/DECT-base. My dial options are rTtWw, automon=*1, blindxfer=## Whether I am calling (to my cell) or being called (from my cell), only the caller can initiate recording or transfer, never the callee... (Which is weird, as I would never ask someone calling me 'please press ##2012' to have a transfer... G) Any hints where to start? -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail and sendmail
Hi, I`m a beginning Asterisk and Sendmail user. I am trying to setup my voicemail to send emails to a certain email address. It doesn't work, and I think I've figured out what it is. There is probably a spam-feature at my provider (that I am using as smart host in sendmail) to not accept emails coming from [EMAIL PROTECTED] If I start a telnet session on port 25 locally and go at it manually, an email with MAIL FROM: [EMAIL PROTECTED] never makes it, while the exact same email with MAIL FROM: [EMAIL PROTECTED] actually gwets to my inbox. How do I make it so that asterisk emails as send using [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Is it an asterisk thing or a Sendmail problem? Because my logs show that the email is send from [EMAIL PROTECTED] Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
