Re: [Asterisk-Users] recall button using tdm400 Australia
Paul == Paul Liew [EMAIL PROTECTED] writes: Paul You are correct - rxflash and flash in zapata does the Paul equivalent, but I should also have said in my earlier post Paul that you need to drop the max pulse time (for pulse Paul dialling) to be less than the hook flash timing. Default Paul settings for max pulse is 150ms, which inteferes with Paul Australian hook flash of 100ms. - It does work, as it is Paul running in our setup here. Arhhh... That makes sense. I suspect you think it is misinterpreting the flash as a pulse used in pulse dialing. Later: I set pulsedial=no in zapata.conf, it doesn't help. That leaves: Paul You need to set rxflash and flash as max and min times for Paul the hookflash to work. I am sorry, you lost me here? You mean set rxflash to the max and flash to the min time? What times should I use? Currently I have: pulsedial=no flash=100 rxflash=100 -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] recall button using tdm400 Australia
Paul == Paul Liew [EMAIL PROTECTED] writes: Paul flash=80 Paul rxflash=120 Unfortunately, that doesn't help. Any more ideas? Thanks. -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] recall button using tdm400 Australia
Hello, How do I get the recall button working on a phone attached to a TDM400 FXS port using Asterisk? I did a web search, and found people with exactly the same problem, but no solution. I suspect the timing is set for American standards, is it possible to get it to work with Australian phones? Thanks. -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] recall button using tdm400 Australia
Paul == [EMAIL PROTECTED] writes: Paul What are you looking to do? Redial the last dialled number Paul or call a missed call? Paul I have a feeling that both items are dealt with by the Paul handsetand the reason the second might not work is due Paul to lack of CALLID. N. RECALL button Australia == FLASH I want to put one call on hold transfer them. -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] recall button using tdm400 Australia
Paul == Paul Liew [EMAIL PROTECTED] writes: Paul Yes you can - change ZT_DEFAULT_FLASHTIME from 750ms to Paul 100ms in zaptel.h Ideally I would like to continue using the pre-built binary for Debian. If possible. Also, I would assume that rxflash and/or flash in zapata.conf does the same thing, but so far I haven't had any luck. As such, I am not entirely convinced changing the source code would help either... When I push recall, I get an interruption in the call, but either the call is disconnected (no, it isn't on hold) and I get the dialtone, or (more likely) I get the same call back again. The fact I get both behaviours seems weird. Perhaps it is a bug in asterisk 1.0.9? -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] recall button using tdm400 Australia
Brian == Brian May [EMAIL PROTECTED] writes: Brian I want to put one call on hold transfer them. read I want to put one call on hold and transfer them. obviously ;-). -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] one outgoing call == one call per minute
On Fri, Nov 11, 2005 at 05:24:44PM +1100, Brian May wrote: ... At first it appears to work perfectly - sound works in both directions, etc. After approx one minute, what happens depends on the remote phone: * if voice mail, the Sipura caller gets transferred to voice mail. * the person at the remote end gets call waiting, and finds the original caller again. The first call is a dead (not hung up) connection. Its almost like the first caller pushed recall, and redialed the number again, but the first caller can hear the second caller's original call, but not vice-versa. This is hard to hear though, since the second caller is usually trying to say hello? hELL? in competition with The problem seems to be something to do with reinvite support, if I turn that off for engin, it seems to work. Curious. I see some comments that reinvite does not always work via NAT, but initially it seems to work fine (for the first 40-60 seconds). Oh well. At least I got it working now. -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] iax2 config sanity check
On Wed, Nov 09, 2005 at 08:31:10PM -0500, asterisk wrote: On the CLI type iax2 debug and then make the call. Paste your results in reply. Thanks for the suggestion. Just the first one: --- cut --- Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 8ms SCall: 1 DCall: 0 [202.91.207.49:0] VERSION : 2 CALLED NUMBER : 5999 CALLING NUMBER : 6003 CALLING NAME: Brian May LANGUAGE: en USERNAME: microcomaustralia FORMAT : 512 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 191595578 --- cut --- Port 0? WTF??? No wonder it isn't working - I don't think 0 is a valid port. Where is 0 coming from? I never specified the port, and didn't think it was required. Using port=4569 in iax.conf solves the problem. Now when I dial the invalid number I immediately get No such context/extension, which is what I expect. Next step - dial a real number ;-). -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] one outgoing call == one call per minute
Hello, I am just in the process of getting asterisk connected to an egin VOIP account in Australia. What works: * ALL incomming calls. * outgoing calls from Zap interface. What almost works: * outgoing calls from Sipura/SPA2100-3.2.1 (eth0) via Asterisk to Egin (eth1). At first it appears to work perfectly - sound works in both directions, etc. After approx one minute, what happens depends on the remote phone: * if voice mail, the Sipura caller gets transferred to voice mail. * the person at the remote end gets call waiting, and finds the original caller again. The first call is a dead (not hung up) connection. Its almost like the first caller pushed recall, and redialed the number again, but the first caller can hear the second caller's original call, but not vice-versa. This is hard to hear though, since the second caller is usually trying to say hello? hELL? in competition with the voicemail message. At the point the problem occured my brother hang up the sipura phone, and contacted me on extension 6003 (as seen in attached log files for eth0). I have attached tethreal logs on eth0 (Sipura to asterisk) and eth1 (asterisk to egin). It would appear that asterisk is generating an extra INVITE command on eth1. Why??? I don't see any similar INVITE command on eth0. I also see SIP Status: 481 Call/Transaction Does Not Exist errors - but this happens after the extra INVITE so I think it is another symptom of the problem as opposed to the problem itself. Disclaimer: I am expert on the SIP protocol... -- Brian May [EMAIL PROTECTED] 0.00 192.168.87.70 - 192.168.87.1 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 0.000518 192.168.87.1 - 192.168.87.70 SIP Status: 407 Proxy Authentication Required 0.007510 192.168.87.70 - 192.168.87.1 SIP Request: ACK sip:[EMAIL PROTECTED] 0.011840 192.168.87.70 - 192.168.87.1 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 0.012223 192.168.87.1 - 192.168.87.70 SIP Status: 100 Trying 6.346837 192.168.87.1 - 192.168.87.70 SIP/SDP Status: 183 Session Progress, with session description 8.067548 192.168.87.1 - 192.168.87.70 SIP/SDP Status: 200 OK, with session description 8.078536 192.168.87.70 - 192.168.87.1 SIP Request: ACK sip:[EMAIL PROTECTED] 8.078722 192.168.87.1 - 192.168.87.70 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5061, with session description 8.090110 192.168.87.70 - 192.168.87.1 SIP/SDP Status: 200 OK, with session description 8.090276 192.168.87.1 - 192.168.87.70 SIP Request: ACK sip:[EMAIL PROTECTED]:5061 49.294879 192.168.87.1 - 192.168.87.70 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5061, with session description 49.309931 192.168.87.70 - 192.168.87.1 SIP/SDP Status: 200 OK, with session description 49.310156 192.168.87.1 - 192.168.87.70 SIP Request: ACK sip:[EMAIL PROTECTED]:5061 54.985854 192.168.87.70 - 192.168.87.1 SIP Request: BYE sip:[EMAIL PROTECTED] 54.986054 192.168.87.1 - 192.168.87.70 SIP Status: 200 OK 57.518254 192.168.87.70 - 192.168.87.1 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 57.518737 192.168.87.1 - 192.168.87.70 SIP Status: 407 Proxy Authentication Required 57.526203 192.168.87.70 - 192.168.87.1 SIP Request: ACK sip:[EMAIL PROTECTED] 57.530554 192.168.87.70 - 192.168.87.1 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 57.530882 192.168.87.1 - 192.168.87.70 SIP Status: 100 Trying 57.553155 192.168.87.1 - 192.168.87.70 SIP Status: 180 Ringing 60.473068 192.168.87.1 - 192.168.87.70 SIP/SDP Status: 200 OK, with session description 60.487031 192.168.87.70 - 192.168.87.1 SIP Request: ACK sip:[EMAIL PROTECTED] 64.044937 192.168.87.70 - 192.168.87.1 SIP Request: BYE sip:[EMAIL PROTECTED] 64.045068 192.168.87.1 - 192.168.87.70 SIP Status: 200 OK 0.00 202.173.153.89 - 202.61.13.40 SIP Request: REGISTER sip:byo.engin.com.au 0.037776 202.61.13.40 - 202.173.153.89 SIP Status: 200 OK(1 bindings) 11.341364 202.173.153.89 - 202.61.13.40 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 11.377687 202.61.13.40 - 202.173.153.89 SIP Status: 100 Trying 11.450946 202.61.13.40 - 202.173.153.89 SIP Status: 407 Proxy Authentication Required 11.451144 202.173.153.89 - 202.61.13.40 SIP Request: ACK sip:[EMAIL PROTECTED] 11.451315 202.173.153.89 - 202.61.13.40 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 11.507928 202.61.13.40 - 202.173.153.89 SIP Status: 100 Trying 17.675080 202.61.13.40 - 202.173.153.89 SIP/SDP Status: 183 Session Progress, with session description 19.395642 202.61.13.40 - 202.173.153.89 SIP/SDP Status: 200 OK, with session description 19.395894 202.173.153.89 - 202.61.13.40 SIP Request: ACK sip:[EMAIL PROTECTED] 19.396328 202.173.153.89 - 202.61.13.40 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 19.517816 202.61.13.40 - 202.173.153.89 SIP Status: 407 Proxy
Re: [Asterisk-Users] iax2 config sanity check
Brian == Brian Capouch [EMAIL PROTECTED] writes: Brian The username and the peer name aren't the same thing. Brian There is some ambiguity floating around as to just how the Brian syntax parses out fully. Brian Use the username, microcomaustralia (ugh. that name is too Brian long) in front of the peer name, Brian e.g. IAX2/[EMAIL PROTECTED]/5999 and see how that works Brian out. Assuming 5999 is the extension you want to reach at Brian the other end. Thanks for you suggestion. Unfortunately the results are exactly the same. If I keep waiting until the timeout, it says: -- Executing Dial(Zap/1-1, IAX2/ivt/5999) in new stack -- Called ivt/5999 Nov 10 12:14:55 WARNING[2644]: chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 202.91.207.49 on IAX2/ivt/1 (type = 6, subclass = 1, ts=3, seqno=0) -- Hungup 'IAX2/ivt/1' or after your suggestion: -- Called [EMAIL PROTECTED]/5999 Nov 10 12:06:32 WARNING[2644]: chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 202.91.207.49 on IAX2/ivt/1 (type = 6, subclass = 1, ts=9, seqno=0) -- Hungup 'IAX2/ivt/1' so it seems to be getting the correct IP address, even though nothing is getting sent. Curious. Brian You shouldn't be pulling your hair out even if nobody answers your Brian emails. Just play around with the different parts of the dialstring Brian and watch the CLI. It's fun. Yes, if this was the only thing I had to do it would be fun ;-). Unfortunately I have to deal with a constant stream of telemarketers wanting to test my new asterisk system out, debugging software problems that deteriorate into hardware problems (so finding the original software bug is impossible[1]), and other real life issues. Notes: [1] no, this wasn't asterisk. -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] iax2 config sanity check
Hello,
Based on my reading and understanding of the documentation, in
extensions.conf all I need is:
exten = _5XXX,1,Dial(IAX2/ivt/${EXTEN})
As asterisk will look up the rest of the configuration in iax.conf:
--- cut ---
[ivt]
username=microcomaustralia
type=friend
host=dynamic
context=default
host=202.91.207.49
permit=0.0.0.0/0.0.0.0
auth=rsa
inkeys=ivt
outkey=microcomaustralia
--- cut ---
However this doesn't work - I get no packets whatsoever getting sent to
202.91.207.49. In fact no packets I have observed look related in
anyway.
Asterisk displays:
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, IAX2/ivt/5999) in new stack
-- Called ivt/5999
[ pause until I hang up ]
-- Hungup 'IAX2/ivt/1'
== Spawn extension (international, 5999, 1) exited
non-zero on 'Zap/1-1'
It seems that I have to include the full IP address and key in the Dial
instruction. Then it works.
From memory if I wait long enough it will timeout, but the timeout error
doesn't help track the problem down.
What am I doing wrong?
I get the impression that it is finding the [ivt] entry in iax.conf, but
unable to prove it beyond doubt. If it can't find it the results appear
to be exactly the same, except the number of the channel in IAX2/ivt/1 is
incremented after every attempt.
Any help before I pull all my hair out would be much appreciated.
Asterisk version 1.0.9.dfsg.1-3
--
Brian May [EMAIL PROTECTED]
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Re: [Asterisk-Users] TDM400 takes Zap/4 line off hook
On Fri, Oct 07, 2005 at 08:13:39AM +1000, Brian May wrote: Anyway, yesterday I noticed I cannot make incoming calls on Zap/4 (I get the busy signal) or outgoing calls (I get silence). I unplugged the phone connection from Zap/4, and tried ringing it, it started ringing. Hello, Just to followup on my previous post - it appears the module on the TDM400 card was faulty - I got it replaced under warranty and it works again now. Thanks -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] fax - conversion problem
asterisk == asterisk [EMAIL PROTECTED] writes:
asterisk The problem is in the tiff2ps, not in the ps2pdf. I
asterisk found that if I remove the -h and -w parameter
asterisk everything is OK
My computer has a tiff2pdf command (from the libtiff-tools Debian
package), so I can do everything in one command:
--- /usr/local/sbin/mailfax ---
#!/bin/sh -e
FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
REMOTESTATIONID=$4
FAXPAGES=$5
FAXRESOLUTION=$6
if [ ! -f $FAXFILE ]
then
echo Fax $FAXFILE not found 2
exit 1
fi
tiff2pdf -pA4 $FAXFILE |
mime-construct --to $RECIPIENT --subject Fax from $FAXSENDER \
--attachment fax.pdf --type application/pdf --file -
--- cut ---
I am not sure of the -pA4 option, but I don't know enough about FAX
standards to change it - it might be better to set the resolution with
-r depending on the FAXRESOLUTION parameter. I did a web search on the
resolution for the different modes, and got numerous different answers
for the same thing :-(.
Oh, and this is called with:
[fax]
exten = s,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${REMOTESTATIONID} ${FAXPAGES} ${FAXRESOLUTION})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,SetVar([EMAIL PROTECTED])
exten = s,3,rxfax(${FAXFILE})
--
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Re: [Asterisk-Users] Can someone please explain caller line identification
On Thu, Oct 20, 2005 at 10:09:01AM +1000, Howard Lowndes wrote: There is/was a patch to * that suggested that in /channels/chan_zap.c the variable DEFAULT_CIDRINGS should be changed from 1 to 2 to suit Australian conditions and I had this done and everything worked. Sorry, this probably won't help you... However, I don't think you need to change the source code, I think you can do the same thing by setting: sendcalleridafter=2 in /etc/asterisk/zapata.conf. At least this works for me. Since I recompiled * I have lost inbound CLID recognition but have gained the distinctive ring recognition ability which I previously didn't have. I would speculate that distinctive ring recognition broke CLID. I also have a Wait(2) at the start of the relevant amswering dial plan as also recommended. I have found I don't need this. The extension file does not appear to be processed until caller-id is established. This would just add extra delay the caller must wait until the phones start ringing. In fact,if you enable callerid without callerid support on the telephone line, I noticed the phones will ring 2+ times before asterisk will detect it - as it is looking for callerid information that isn't there. So I am curious - how long is it from when the phone line starts ringing to when asterisk detects the call? I have only had some minor issues with callerid and related stuff: * on one phone it now indicates all phone calls are OUT OF AREA. The phone number appears OK. I can't work out where this message comes from. * one another cordless phone system, it registers when there is a message waiting, but it never resets back to no message waiting, even after the message has been deleted. (Note this is using a Sipura POTS adaptor). I have: └─(13:21!505:%)── dpkg -l asterisk ──(Thu,Oct20)─┘ Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed |/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err: uppercase=bad) ||/ Name VersionDescription +++-==-==- ii asterisk 1.0.9.dfsg.1-3 open source Private Branch Exchange (PBX) -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] TDM400 takes Zap/4 line off hook
Hello, I have a TDM400 with 2 FXS modules and 2 FXO modules, as follows: Zap/1 - internal phone Zap/2 - internal modem Zap/3 - exchange Zap/4 - exchange Recently I upgraded zaptel (and kernel modules) from 1.0.9-1.1 to 1.0.9.1 and upgraded asterisk to 1.0.9.dfsg.1-3, which are Debian versions for Debian/sarge of the packages I believe the Debian maintainers created (not in Debian) The above upgrade may not be related. It might be coincidence. Anyway, yesterday I noticed I cannot make incoming calls on Zap/4 (I get the busy signal) or outgoing calls (I get silence). I unplugged the phone connection from Zap/4, and tried ringing it, it started ringing. I plugged it back in, and my computer died. If a picked of the phone on Zap/1 all I got was extensive static noises. Resetting my computer didn't help. Turning it off and on didn't work. Eventually I got it going again. When the computer booted, it didn't detect the zaptel card for some reason. It appears that asterisk will lock the computer up if it loads and it can't detect the card for some reason. This is confusing, because when it crashes something else is happening in the foreground (usually loading proxy server). However, if I stopped asterisk from automatically loading the computer would boot again. In the process I removed the TDM400 card, this only confirmed my theory. Anyway, at the moment everything is working again at the state it was when I started. Any ideas on how to make Zap/4 work correctly? Zap/3 still appears to work fine. -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] call parking timeout
On Wed, Aug 24, 2005 at 09:04:53AM -0500, Eric Wieling aka ManxPower wrote: You looked at the features.conf.sample file? Yes. I don't see how that helps, at least in my version. There is a parameter to change the timeout time, but I don't want to change the time, I just want to change the behaviour when this timeout is exceeded and the default behaviour doesn't work. Or are you talking about some CVS-only feature here? -- Brian May [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] call parking timeout
Hello, If a parked call times out, it will ring the original extension again. If this fails for any reason (e.g. the extension is busy), then the caller receives silence until the timeout time, and then gets the busy signal and is disconnected. I consider disconnecting callers like this very unprofessional, and could happen for instance if you receive a new call when you just parked the previous call (especially if you have caller-waiting disabled). Is there any solution to this? Ideally I would like it to park the call again, either if no answer within X seconds or if the extension is busy. An alternative would be to forward the call to another extension and/or voice mail. Thanks in advance. -- Brian May [EMAIL PROTECTED] ___ 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
