Re: [Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX
Llorenç Suau [EMAIL PROTECTED] writes: Any suggestions, to how I can make that the PBX receives correctly the call, PREFIX+number, to make the external call. Does this link have the right to make calls to the outside world on the PBX? Normally this feature is turned off on typical PBX. cu, Wolfgang ___ --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] Modems dialing over sangoma a104d
Rich Adamson [EMAIL PROTECTED] writes: Sean Cook wrote: I have a sangoma 104d that is our main pbx now( legacy system died ). I have replaced every phone in the building and things are going very well. We have fax working well and calls are routing properly... All is well... Except for our support modems... we have support people that dial out with modems across our PRI's. These modems are attached to an Adtran 750 with 24 FXS's. I have disabled echo cancelation on the T1 that is connected to the Adtran but negotiation is still really rough. I am bridging across the same card and it isn't doing very well... has anyone done this with reasonably successful results? I am not looking for 56K I am looking for around 9600 to 14.4.. Can we assume that you've got the correct timing parameters set on the 104d? (eg, are you sync'ing your 104d from the telco?) If not, get that corrected first as it makes a major difference with modem calls. That's the point! We had the same issues: modem calls dropping after a few minutes. Get at least wanpipe-beta7-2.3.4.tgz and set the reference clock to the telco line and set MASTER to this line. cu, Wolfgang ___ --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 and Siemens Legacy PBX
James Arscott [EMAIL PROTECTED] writes: Hi Small progress, though combining the suggest below, enabling overlapdial and a few other things I have got the following : When you hit 9 on the simenes, you hear a dial tone. As soon as you hit another number to start dialling it complains with some generic error on the siemens handset. What I see from asterisk at the same time [...] -- Starting simple switch on 'Zap/62-1' -- Accepting overlap call from '697000' to 'unspecified' on channel 0/31, span 2 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: INFORMATION (123) [70 02 81 39]LI Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9' ] -- Processing IE 112 (cs0, Called Party Number) Hmm? That's not overlap dialing. You get the complete called number in one single Message. -- Processing IE 112 (cs0, Called Party Number) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap Receiving, peerstate Overlap sending Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 81]LI Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] ... and Asterisk's answer is: Unallocated number. It seems your Siemens PBX doesn't do it right. We had some issues with other PBXs and Asterisk when Asterisk was the NET-side. Try to reverse the roles, so that Siemens is NET and Asterisk CPE. That helped here with an Alcatel. cu, Wolfgang ___ --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] HFC-S Cards in the UK
Ron Wellsted [EMAIL PROTECTED] writes: So while there HFC cards out there, it seems that they are going to get harder to find. We got a few of these from Conrad. They are in Germany and I am not sure if this one is the same as ours. But at EUR 24.95 per card you cannot loose to much. http://www.conrad.de/goto.php?artikel=955078 hth, Wolfgang ___ --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 and Siemens Legacy PBX
James Arscott [EMAIL PROTECTED] writes: Asterisk is not matching the extension from the siemens because the siemens has not even sent one yet, it is still waiting for a dial tone. When I hit 9 on the siemens it does not get a dial tone from asterisk, I assume this is because I have not told asterisk to give it one(dur!) How should I tell asterisk how to handle this, I have defined it a context and I know its making it this far, but I don¹t know how to get the next bit coded. Any help appreciated ! Have you turned on overlapdial in zapata.conf? Most PBX's send numbers in overlap mode, yours does it also. Next, you should add something like this to your Siemens-context: exten = s,1,WaitExten(2) This waits for the numbers which should now come from the Siemens side. hth, Wolfgang ___ --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 and Siemens Legacy PBX
James Arscott [EMAIL PROTECTED] writes: My concern is still over if L2 and L3 are Œup¹ on my ISDN between the asterisk and siemens, I do my settings look right ? I thought my timing on span 2 maybe incorrect ? If you do a pri show span 2 you get a short info about the state of the span. Next, you can try pri debug span 2 and watch the ISDN conversations. This should give you the info if your link is set up correctly. cu, Wolfgang ___ --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] Yet another problem with incoming SIP calls and 407
Eric \ManxPower\ Wieling [EMAIL PROTECTED] writes: Wolfgang Zweimueller wrote: Hi all, when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the caller has a username in it's From-Address which also exists in my sip.conf then my system answers with 407 Proxy Authentication Required. If it's nonexistent username then callin works fine! It seems that this is a problem in the SIP implementation of Asterisk and found a few hints on how to resolve this (allowguest=yes, insecure=invite,port etc.). But none of them does help! Can anyone suggest what I else could try? in sip.conf [general] context=INVALID Then put the correct context= line for each sip user/friend/peer. Unauthenticated calls use the options in [general] That's already there! Any other ideas? cu, Wolfgang ___ --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] Yet another problem with incoming SIP calls and 407
Hi all, when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the caller has a username in it's From-Address which also exists in my sip.conf then my system answers with 407 Proxy Authentication Required. If it's nonexistent username then callin works fine! It seems that this is a problem in the SIP implementation of Asterisk and found a few hints on how to resolve this (allowguest=yes, insecure=invite,port etc.). But none of them does help! Can anyone suggest what I else could try? Thanks, Wolfgang ___ --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] zapata.conf: recent changes?
Mimmus [EMAIL PROTECTED] writes: Hi, after a few of upgrades, I noticed these messages in full debug log: Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring internationalprefix Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring nationalprefix Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring localprefix Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring privateprefix Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring unknownprefix Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring priindication Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring facilityenable Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring signalling A lot of changes? No, IMHO does it appear when you issue a reload command on the CLI. Because this options need a complete *-restart. cu, Wolfgang ___ --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] E1 hardware for asterisk
Tristan [EMAIL PROTECTED] writes: This server ( an IBM X-Series 346 dual 3Ghz Xeon with 2gb ram ) has to serve about 60-70 incoming/outgoing PSTN simultaneous calls ( IVR and max 30-40 conferences... ) and about 10-20 SIP calls to begin... I tried the Quad-port Digium cards in this special machine and it crashed the machine! I don't know why it happened but it did it again and again. Now I have a Sangoma A104 and it looks O.K. There is an issue with CPU-load but as far as I can see it's no problem. We will connect this machine to the telco line the next days. cu, Wolfgang ___ --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] E1 hardware for asterisk
Hi Alex, Asterisk [EMAIL PROTECTED] writes: Wolfgang: What kind of CPU-load issue on A104? Could you give me a link or something? I also use A104 and it works very good, but recently I noticed a behavior which is maybe connected with this issue, so more info would be very helpful for me :) There's a thread regarding this issue in this Mailing list. First Msg-id is: [EMAIL PROTECTED], Subject is: Experience with IBM X346 machines and Sangoma This machine is not connected to any line at the moment. So I can not say if this issue makes a real problem. I can give more info next week. What did you observe on your machine? cu, Wolfgang ___ --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] Experience with IBM X346 machines and Sangoma
Rich Adamson [EMAIL PROTECTED] writes: Cosmin Prund wrote: I wanted to see where those periodical spikes are coming from so I started shutting things down. The first thing to go was Asterisk. [...] Is there something funny happening with my zaptel? Wolfgang Zweimueller, can you give this a try too? Does your spiking stop when you stop zaptel? The spikes go away after unloading wanrouter modules but *before* removing the zaptel module. Seems I have to contact Sangoma. Another nice issue: after removing the af_wanpipe and the wanpipe module the machine crashed :-( There have been multiple threads over the last two years about the exact same 'vmstat 1' results, and no one has ever come up with a logical explanation as to why it occurs. Well, drivers evolve over the years and things can get better ;-) And I wanted to know if there is a solution for this special machine. Of the several (probably hundreds) of posts in the past, it does not seem to be a linux distro issue, and stopping zaptel always removes the symptom. I am also pretty sure that it is not the distro. I have Debian with non-debian kernel. It seems the majority of folks that were involved with this in the past 'assumed' the results were what was impacting fax through the TDM400. But, don't think anyone proved that. Dont't know anything about TDM400 but we had some issues with Modems which were using the Asterisk-path. No other guesses at this time. I got a mail from David Elbel. He suggested to recompile zaptel drivers *after* installing the Sangome drivers. But that did not help. cu, Wolfgang ___ --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] Experience with IBM X346 machines and Sangoma
Hi All, I have read many posts about problems with Asterisk on some systems. I also set up Asterisk on many different boxes. But I have never seen the following... There is an IBM X346 (3.4GHz Xeon) with one Sangoma A104. This system is currently idle, that means there is nothing running except Asterisk (1.2.7.1). We are handling no calls now, but if I do a vmstat, I get peaks in system load up to 40%! Here is an example: procs ---memory-- ---swap-- -io --system-- cpu r b swpd free buff cache si sobibo incs us sy id wa 0 0 0 375080 161660 14223200 0 0 4160 187 0 4 96 0 0 0 0 375080 161660 14223200 0 0 4251 207 0 1 98 0 0 0 0 375080 161660 14223200 0 0 4205 179 0 9 92 0 0 0 0 375080 161660 14223200 036 4151 217 0 3 97 0 0 0 0 375080 161660 14223200 0 0 4026 187 0 0 100 0 0 0 0 375080 161660 14223200 0 0 4042 205 0 14 86 0 0 0 0 375080 161660 14223200 0 0 4019 184 0 38 63 0 0 0 0 375080 161660 14223200 0 0 4062 208 0 0 100 0 0 0 0 375080 161660 14223200 0 0 4028 196 0 2 99 0 0 0 0 375080 161660 14223200 016 4075 223 0 19 81 0 1 0 0 375080 161660 14223200 0 0 4029 197 0 0 100 0 0 0 0 375080 161660 14223200 0 0 4043 199 0 1 99 0 0 0 0 375080 161660 14223200 0 0 4045 194 0 6 94 0 0 0 0 375080 161660 14223200 0 0 4032 196 0 24 77 0 0 0 0 375080 161660 14223200 012 4045 212 0 0 100 0 0 0 0 375080 161660 14223200 0 0 4028 188 0 0 100 0 In contrast to the above I have a Dell 2850 running Asterisk and a lot of other things (but no PRI card). This box is (according to vmstat) almost always 100% idle! Is anyone running a similar X346-system? What is the load and how does Asterisk behave on it? Can anyone explain what is happening here? Thx, Wolfgang ___ --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, QSIG and Tenovis PBX?
Hi Johann, Johann Hanne [EMAIL PROTECTED] writes: Hi, we are still trying to properly connect a Tenovis PBX to an Asterisk server (asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this time with QSIG. I tried with asterisk 1.2.2 against Alcactel 4400 on Monday. We had partially success. But at a specific config on the Alcatel side, the called number was not set by the SETUP message but via INFORMATION messages. Well, libpri doesn't like it this way. AFAIR, libpri does Q.SIG basic call, so you should set the Tenovis also to basic call. If this doesn't help, please run a pri debug span 1 while you make calls and post the output. My conclusion with Q.SIG: do not use it at this implementation level. YMMV. cu, Wolfgang ___ --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] Howto cut the first digit
Christian Reelfs [EMAIL PROTECTED] writes:
Hi, sorry for this noop question,
but does anybody know how to cut the first digit of a variable?
example:
044612345
should be after cut operation:
44612345
Look at README.variables! It says:
,
| The format for removing characters from a variable can be expressed as:
|
| ${variable_name[:offset[:length]]}
|
| If you want to remove the first N characters from the string assigned
| to a variable, simply append a colon and the number of characters to
| remove from the beginning of the string to the variable name.
|
| ;Remove the first character of extension, save in number variable
| exten = _9X.,1,Set(number=${EXTEN:1})
`
cu,
Wolfgang
___
--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, QSIG and Tenovis PBX?
Dinesh Nair [EMAIL PROTECTED] writes: On 03/31/06 19:49 Wolfgang Zweimueller said the following: My conclusion with Q.SIG: do not use it at this implementation level. YMMV. i'll beg to differ. we've used Q.SIG successfully with an Ericsson MD110 for a customer in thailand. Well, that's the YMMV. I have it also running with an Alcatel 4200. But my last experience with the 4400 showed me that there is something missing in the Q.SIG implementation. I also have seen some weird things with Q.SIG on BRI. And as long as I don't know what will happen when I connect * to some PBX, I won't tell my customers about Q.SIG. cu, Wolfgang ___ --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] Working Asterisk with Austrian ISDN p2p
Marcus Hofbauer [EMAIL PROTECTED] writes:
Hi!
I'm looking for someone who has successfuly setup an asterisk in
austria with isdn in p2p mode and chan_capi.
There is is a special problem in austria with DID. If someone is
dialing the phone number of the asterisk pbx like 12345-0, zero is
passed as an DID, but in Austria u can dial 12345, and the DID which
is passed is empty.
It seems that asterisk cant handle this. Any ideas?
I am doing this (with zaphfc) and it works. I have it running with DID
and overlap dial. It's not a big deal. Create the s-Extension and
use WaitExten(). Then create DID-extensions.
Here is an example:
exten = s,1,DigitTimeout(5)
exten = s,2,WaitExten(3)
exten = s,3,Macro(dialone,SIP/${DURCHWAHL},10)
exten = 0,1,Macro(dialone,SIP/${DURCHWAHL},10)
exten = 10,1,Macro(dialone,SIP/${DURCHWAHL},10)
hth,
Wolfgang
___
--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] Austria isdn p2p empty DID
Marcus Hofbauer [EMAIL PROTECTED] writes: BUT ... If someone is dialing the PBX head number without any extension, asterisk can't handle this ... the DID in this case is empty Any ideas how to handle this? Try the WaitExten application. cu, Wolfgang ___ --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] Alcatel 4200 series pbx
Igor Neves [EMAIL PROTECTED] writes: Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? Hi Igor, we are doing that. Bristuffed Asterisk with two HFC-cards is running as NT and the Alcatel is CPE. Do you have any specific problem? cu, Wolfgang ___ --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] QSIG error -- can somebody explain?
Johann Steinwendtner [EMAIL PROTECTED] writes: I can only guess, but I think I can remember that the creflen needs to be 2 octets for qsig. Check what the Alcatel switch sends in the setup message to *. Thanks, I will have a look at that. Anyway, why do use QSIG ? Does name display work on the * implementation ? It is not because of name display but of an issue with call routing on this PBX. We have a running setup with Euro-ISDN. If we can switch over to Q.SIG there would be a benefit for the customer. Best regards Hans P.S.: Schoene Gruesse an Kurt Krenn Wir gemacht! cu, Wolfgang ___ --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] QSIG error -- can somebody explain?
Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way: switchtype=qsig signalling = bri_cpe facilityenable = yes My experience now is that it is possible to signal a call (both outgoing and incoming) but as soon as the callee takes off the hook the call-setup crashes. Below is the debug log of an outgoing call to a service number of the telco which tells the current time. (The point is that the called number immediately answers the call.) As you can see the Alcatel side answers to our SETUP message with a RELEASE COMPLETE and a cause number 100. This cause (taken from ECMA-143) means: Invalid information element contents , | This cause indicates that the equipment sending this cause has received an | information element which it has implemented; however, one or more of the fields | in the information element are coded in a way that has not been implemented by | the equipment sending this cause. ` Can somebody explain what the problem is? Configuration error, a bug, a problem on the Alcatel-side? Thanks in advance, Wolfgang -- Executing Dial(SIP/1993-567b, Zap/g1/006621503|55|j) in new stack 1 -- Making new call for cr 136 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (8) len=32 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 891 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 061 211 801 311 391 391 331 ] 1 Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1993' ] 1 [1 701 0a1 c11 301 301 361 361 321 311 351 301 331 ] 1 Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '006621503' ] -- Called g1/006621503 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: (null) (100), class = Protocol Error (6) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null 1 No response to SETUP message 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending 1 Protocol Discriminator: Q.931 (8) len=8 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: DISCONNECT (69) 1 [1 081 021 811 921 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (18), class = Normal Event (1) ] -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Request, peerstate Disconnect Indication -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Busy(SIP/1993-567b, ) in new stack == Spawn extension (dialout, 436621503, 102) exited non-zero on 'SIP/1993-567b' 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 d11 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null ___ --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
