Re: [asterisk-users] Asterisk dialplan date and time operations
Thanks!
I got it now!
Here is a sample for a delayed callback after a caller gets to a users
voicemailbox. Purpose: Reminder for people that they got a message on their
v. box.
exten = 1002,1,Answer
exten = 1002,2,Set(CHANNEL(musicclass)=default)
exten = 1002,3,Queue(test|t|||5)
exten = 1002,4,Voicemail(b1205)
exten = 1002,5,System(echo -e Channel: SIP/we-static\\nCallerID:
VOICEMAIL
8500\\nContext: test\\nExtension: 444 /tmp/${UNIQUEID}.call)
; add 15 minutes (in seconds 900) to the epoch time
exten = 1002,6,Set(newepoch=${MATH(${EPOCH} + 900 |int)})
; write it out for debugging purpose
exten = 1002,7,NoOp(${newepoch})
exten = 1002,8,System(touch -t ${STRFTIME(${newepoch},,
%Y%m%d%H%M)} /tmp/${
UNIQUEID}.call)
exten =
1002,9,System(mv /tmp/${UNIQUEID}.call /var/spool/asterisk/outgoing
/)
exten = 1002,10,Hangup
Kind Regards,
Erik
Am Mittwoch, 2. Januar 2008 18:59 schrieb Tilghman Lesher:
On Wednesday 02 January 2008 09:34:24 Erik Wartusch wrote:
No it's even simpler. ( I dont need an IF case)
I just want to add e.g. 15 minutes to the current date / time:
So simply said:
${STRFTIME(${EPOCH},,%Y%m%d%H%M)} + 15 minutes!
My question was how can I do that.? Of yourse e.g. if it's 23.57 pm and I
add 15 minutes the day should increase +1 and the hours start with 0:x
the minutes with 12 ( and not 72 as the normal addition would result).
${STRFTIME($[${EPOCH} + (15 * 60)],,%Y%m%d%H%M)}
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to automaticaly close callswhenAsterisk didn't receive the bye request ?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B From: Jared Smith [EMAIL PROTECTED] There is a SIP timers patch in the bug tracker (see http://bugs.digium.com/view.php?id=10665) that currently implements this, and it's being tested in the team/group/sip_session_timers/ branch in SVN. Please test this out and help provide feedback, so that we can get this put into Asterisk in time for the next major release. Jared, I would think of using rtptimeout. There is a reason why you did not mention it and I am curious as to why. Does rtptimeout help if you are using canreinvite=yes ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to automaticaly close calls whenAsterisk didn't receive the bye request ?
The rtptimeout feature has a few limitations: . It is ineffective when the RTP is not terminated on Asterisk. . It can cause false session hangups if the remote SIP UA does not support silence suppression . The companion rtpholdtimeout can cause false hangups if you make incorrect judgment on how long a call hold can last. . The rtptimeout period is not negotiated throughout the SIP signaling path i.e. between the UAC, UAS, and intermediary proxies. So it does not help clear the session state throughout the network (when your BYE doesn't make it to all the entities in the SIP signaling path). The SIP session-timers feature addresses all of the above limitations. -- Raj Jared, I would think of using rtptimeout. There is a reason why you did not mention it and I am curious as to why. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lamps on Snom phones
Phil Knighton schrieb: Thanks for your reply Stefan :-) I'm using Asterisk 1.4.10 now, I was using 1.2.16. My config hasn't changed between the two, both had the hints set in extensions.conf with entries such as exten = 510,hint,SIP/510. Each of the Snom phones has function keys programmed with the relevant phones to monitor, with a function key set to Extension (Destination on older Snoms) and sip:[EMAIL PROTECTED];user=phone - for example. The keys will still call the programmed phone when pressed, but that is the only time the lamp works - when physically pressed. If I dial 510 manually, the 510 function key used to flash to show it was ringing. This no longer happens. This was all working fine on 1.2, but since moving to 1.4 all of the lamps on all of the phones have stopped working! I've tried the points mentioned in a previous answer with no luck, as far as I can see form the examples in extension.conf (v1.4) I have configured the hints correctly. I've also checked the sip.conf file, and set the subscription settings as follows: allowsubscribe=yes subscribecontext=softoption-hints notifyringing=yes notifyhold=yes limitonpeers=yes What next? well, what do the commands core show hints and sip show subscriptions tell you? Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Hello, 2008/1/1, Steve Underwood [EMAIL PROTECTED]: Hi Rob, Rob Hillis wrote: Well that answers that question. I see that t38modem provides an H232 modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact that it requires a kernel recompile on most newer distros.) Steve Underwood wrote: Rob Hillis wrote: Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know. There are actually two reasons. One is that T.38 over IAX is not defined. The other is the current T.38 termination support in spandsp is only for the full FAX machine it contains. T.38 termination to the class 1 FAX modem (T.31) interface for HylaFAX is a work in progress. When that is done, I hope we will have a sipmodem to replace iaxmodem, offering bother audio and T.38 to HylaFAX functionality. Steve The most recent versions of t38modem can apparently provide both a SIP and H.323 T.38 to class 1 FAX modem interface for HylaFAX. What it cannot provide is an audio FAX interface. What is an audio FAX interface ? I'm not sure to understand what it is. Cheers The sipmodem code I am working on will integrate audio and T.38 FAX processing in a single SIP entity. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom VLAN
This is the whole point behind using VLAN on the phone. Tagged VLAN for your phone with QoS configured accordingly on your switch and untagged VLAN for your PC, both on the same wire. This way you can always guarantee enough bandwidth for your VoIP packets. Thanks, Wojtek On 2-Jan-08, at 1:04 PM, Alex Balashov wrote: On Wed, 2 Jan 2008, Jeremy Mann wrote: Just curious, if I have my Polycom IP 550 phone VLAN tag 30, will the packets I send from my PC(on the PC port of the phone) have the same VLAN tag? THe PC is sending untagged packets. According to this -- http://www.polycom.com/common/documents/whitepapers/ vlans_and_polycom_soundpoint_ip_desktop_ip_telephones.pdf If a PC is connected to the phone, all packets generated by the PC will be passed through unmodified, regardless of the presence of an 802.1q/p tag or its contents. Since PCs do not typically tag frames, this means they will be on the native VLAN. Cheers, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GSM Gateway behind SIP ATA?
I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how do i do this in Asterisk? Basically Asterisk should dial the extension number and then send required number as DTMF tones to the Gateway through the ATA. I am using FreePBX, which allows me to create a custom trunk for the outgoing calls. Hope this could work :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lamps on Snom phones
Sorry, should have put that in my reply. core show hints shows me all the hints as I would expect to see them, for example: [EMAIL PROTECTED]: SIP/510 State:Idle Watchers 6 and core show subscriptions also shows me all the subscriptions, again as I would expect to see them (here you can see the subscription on MY phone for 510): 10.0.0.77phil3c267009a12 [EMAIL PROTECTED] Idle dialog-info+xml none As far as I can tell, Asterisk is setup as it should be but the Snoms just don't pick anything up. I've incorporated the kind responses from other list members, such as setting call limits but to no avail! I've checked the function key settings on the Snom, and adjusted it to match the suggestion from another list member - nothing. Not one lamp on any of the phones will work, and I'm completely baffled as to why. -Original Message- From: Stefan Guenther [mailto:[EMAIL PROTECTED] Sent: 03 January 2008 12:21 To: Phil Knighton Cc: asterisk-users@lists.digium.com Subject: Re: Lamps on Snom phones Phil Knighton schrieb: Thanks for your reply Stefan :-) I'm using Asterisk 1.4.10 now, I was using 1.2.16. My config hasn't changed between the two, both had the hints set in extensions.conf with entries such as exten = 510,hint,SIP/510. Each of the Snom phones has function keys programmed with the relevant phones to monitor, with a function key set to Extension (Destination on older Snoms) and sip:[EMAIL PROTECTED];user=phone - for example. The keys will still call the programmed phone when pressed, but that is the only time the lamp works - when physically pressed. If I dial 510 manually, the 510 function key used to flash to show it was ringing. This no longer happens. This was all working fine on 1.2, but since moving to 1.4 all of the lamps on all of the phones have stopped working! I've tried the points mentioned in a previous answer with no luck, as far as I can see form the examples in extension.conf (v1.4) I have configured the hints correctly. I've also checked the sip.conf file, and set the subscription settings as follows: allowsubscribe=yes subscribecontext=softoption-hints notifyringing=yes notifyhold=yes limitonpeers=yes What next? well, what do the commands core show hints and sip show subscriptions tell you? Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
On Thursday 03 January 2008 15:28:15 Remco Barendse wrote: I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how do i do this in Asterisk? Basically Asterisk should dial the extension number and then send required number as DTMF tones to the Gateway through the ATA. Basically Grandstream HT286 is a single port FXS ATA. In order to interconnect GSM gateway one would need FXO. Are you sure it gives you new dialing tone or this is the * itself you hear? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recorded calls skipping
Steve Totaro wrote: Jay Moore wrote: Greetings, List. I'm having a problem where my recorded calls are skipping every 4-5 seconds are so. I can hear the caller (or callee) just fine and then a second or so of silence followed by the person talking again. I'm saving my calls as .gsm files and it's worked fine for the past 11 months. I make sure I remove the recorded files from my Asterisk box and put them onto our fileserver, so it's not an issue of disk space. No other settings have been changed, so I'm not sure why my calls aren't being recorded properly now. Any thoughts? Thanks in advance, Jay You do not mention call volumes or simultaneous calls being recorded. If you are pushing around 70 or so simultaneous calls then you probably have an I/O issue with your hard drive. Although, I received complaints from the phone users about audio chopping before the recording were affected. I assume you are using the monitor app? Thanks, Steve Totaro At absolute maximum, we're probably recording 7-8 simultaneous calls, but most of the time it's 1-2. It's a newer rig, so I'm more inclined to think it's software and not hardware. Unfortunately, it's an older version of Asterisk, but I've had zero problems until the skipping calls, and if it ain't broke, don't fix it, right? :) I restarted Asterisk and it seems to have solved the problem -- for now at least. I'm a rookie when it comes to Asterisk, any suggestions on what to do to if it happens again? Thanks, Jay ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF trouble
Does anybody have an idea where I can start looking to fix this? Or is this regular behaviour of asterisk that it does not show an extension as busy when it initiated the call? Thanks, Lars -- Zymurgy's Law of Volunteer Labor: People are always available for work in the past tense. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
On Thu, 3 Jan 2008, Benchev wrote: Basically Grandstream HT286 is a single port FXS ATA. In order to interconnect GSM gateway one would need FXO. Are you sure it gives you new dialing tone or this is the * itself you hear? Yes, i am positive that i get a new dialtone from the GSM Gateway. If i dial DTMF codes from a SIP phone connected to Asterisk, i can see the digits appear in the display of the GSM Gateway. But it is a bit incovenient to call an internal extension, wait for the dialtone and then punch in all the numbers of the cell phone i need to call. I would prefer Asterisk to decide where / how to route the call and send the DTMF inband to the ATA device. Thanks!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HFC-S zap channels always busy
Hi list, Attempting to get an ISDN-BRI line connected using an HFC-S PCI card together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch system, I find that I can't access the card's resources because the channels are always be busy. An attempt to call out results in the following CLI output: == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f3698, Zap/g0/[EMAIL PROTECTED]||r) in new stack [Jan 3 15:32:06] WARNING[9769]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081f3698' status is 'CONGESTION' == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down Hopefully, someone here with more experience can point me in the direction of a solution. Here are hopefully some more clues: # lsmod | grep zap zaphfc 13660 1 vzaphfc24984 1 zaptel185956 9 xpp,zaphfc,vzaphfc crc_ccitt 2560 1 zaptel # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) 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: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) It looks like the vzaphfc module creates a virtual interface. I have only one HFC-S PCI card installed. Each channel is (In use) immediately after Asterisk is started. CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefault en default 1from-pstn en default 2from-pstn en default 4from-pstn en default 5from-pstn en default CLI zap restart Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Jan 3 15:40:06] WARNING[9797]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Jan 3 15:40:06] ERROR[9797]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Jan 3 15:40:06] ERROR[9797]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Jan 3 15:40:06] WARNING[9797]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! Not a good idea, because that results in... CLI zap show channels Chan Extension Context Language MOH Interpret the channels disappearing altogether. However, I can restore the situation back to its original, albeit useless, state if I stop and start Asterisk. My configuration files are as follows: /etc/asterisk/zapata-channels.conf (after running genzaptelconf -sd -c nl): group=0,11 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 group= context=default group=0,12 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 group= context=default /etc/asterisk/zapata.conf (supposed to work in the Netherlands): [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=dynamic prilocaldialplan=local nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=100 rxgain=4.5 txgain=-3 group=1 callgroup=1 pickupgroup=1 immediate=yes #include zapata-channels.conf Abbreviated /etc/asterisk/extensions.conf: [globals] [general] [isdn-out] exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r) [internal] exten = 1000,1,Verbose(1|Extension 1000) exten = 1000,n,Dial(SIP/1000,30) exten = 1000,n,Hangup() [phones] include = internal include = isdn-out Any ideas? TIA, Jaap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S zap channels always busy
On Thu, Jan 03, 2008 at 04:08:10PM +0100, Jaap Winius wrote: Hi list, Attempting to get an ISDN-BRI line connected using an HFC-S PCI card together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch system, I find that I can't access the card's resources because the channels are always be busy. An attempt to call out results in the following CLI output: == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down -- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f3698, Zap/g0/[EMAIL PROTECTED]||r) in new stack [Jan 3 15:32:06] WARNING[9769]: app_dial.c:1130 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/1000-081f3698' status is 'CONGESTION' == Primary D-Channel on span 1 down == Primary D-Channel on span 2 down What is the output of: pri show spans (Yes, it is pri and not bri). Do incoming calls work? Hopefully, someone here with more experience can point me in the direction of a solution. Here are hopefully some more clues: # lsmod | grep zap zaphfc 13660 1 vzaphfc24984 1 zaptel185956 9 xpp,zaphfc,vzaphfc crc_ccitt 2560 1 zaptel Interesting... which one of those two is used? I suspect vzaphfc is loaded automatically by udev, unless you have zaphfc explicitly in /etc/modules . # cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) 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: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] AMI/CCS 4 ZTHFC1/0/1 Clear (In use) 5 ZTHFC1/0/2 Clear (In use) 6 ZTHFC1/0/3 HDLCFCS (In use) It looks like the vzaphfc module creates a virtual interface. I have only one HFC-S PCI card installed. Each channel is (In use) immediately after Asterisk is started. CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefault en default 1from-pstn en default 2from-pstn en default 4from-pstn en default 5from-pstn en default CLI zap restart This will not work with digital spans. Try restarting asterisk. e.g: asterisk -R restart now Destroying channels and reloading zaptel configuration. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found [Jan 3 15:40:06] WARNING[9797]: chan_zap.c:1081 zt_open: Unable to specify channel 1: Device or resource busy [Jan 3 15:40:06] ERROR[9797]: chan_zap.c:7501 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 [Jan 3 15:40:06] ERROR[9797]: chan_zap.c:12266 build_channels: Unable to register channel '1-2' [Jan 3 15:40:06] WARNING[9797]: chan_zap.c:11554 zap_restart: Reload channels from zap config failed! Not a good idea, because that results in... CLI zap show channels Chan Extension Context Language MOH Interpret the channels disappearing altogether. However, I can restore the situation back to its original, albeit useless, state if I stop and start Asterisk. My configuration files are as follows: /etc/asterisk/zapata-channels.conf (after running genzaptelconf -sd -c nl): group=0,11 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 group= context=default group=0,12 context=from-pstn switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 group= context=default /etc/asterisk/zapata.conf (supposed to work in the Netherlands): [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=dynamic prilocaldialplan=local nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=100 rxgain=4.5 txgain=-3 group=1 callgroup=1 pickupgroup=1 immediate=yes #include zapata-channels.conf Abbreviated /etc/asterisk/extensions.conf: [globals] [general] [isdn-out] exten =
Re: [asterisk-users] GSM Gateway behind SIP ATA?
Remco Barendse wrote:
I have an analog GSM Gateway that is connected to a normal SIP ATA device.
Basically what it does is this : when you call the extension nr. of the
SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia
a Grandstream HT286.
I would like to use the GSM Gateway to route my outbound cellular calls,
how do i do this in Asterisk? Basically Asterisk should dial the extension
number and then send required number as DTMF tones to the Gateway through
the ATA.
I am using FreePBX, which allows me to create a custom trunk for the
outgoing calls. Hope this could work :)
This should work:
context out-gateway {
_X. {
Dial(SIP/gateway,30,M(dial-gateway^${EXTEN}));
}
}
macro dial-gateway(number) {
Wait(1);
SendDTMF(${number});
}
You dial to gateway, and execute macro upon answer (if i remember
correctly, it should be executed within dialed channel), so macro sends
the number you need to dial on GSM gateway in DTMF, and after that
bridges the call. You might try removing the Wait(1), but your GSM
gateway could expect some idle time before receiving digits so i put it
there.
Regards,
Atis
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lamps on Snom phones
Hello Phil, please check the following details in your asterisk configuration and on your phones. These are the settings that work for me: sip.conf [general] limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes useclientcode=yes canreinvite=yes [user1] secret=user1 host=dynamic username=user1 callerid=user1 97 dtmfmode=rfc2833 context=local type=friend callgroup=1 pickupgroup=1 qualify=yes vmexten=80297 call-limit=20 subscribecontext=local extensions.conf exten = 97,hint,SIP/user1 exten = 98,hint,SIP/smguenther On the SNOM phones: Support broken Registrar: ON Use user:phone: OFF Filter Packets from Registrar: OFF Function Key P6: ACTIVE / EXTENSION / sip:[EMAIL PROTECTED] Hope that helps, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HFC-S zap channels always busy
Quoting Tzafrir Cohen [EMAIL PROTECTED]: What is the output of: pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active Do incoming calls work? I haven't configured that yet. Interesting... which one of those two is used? Good question. I've wanted to test that, but they're all the same: in use. I suspect vzaphfc is loaded automatically by udev, unless you have zaphfc explicitly in /etc/modules . It's not mentioned in /etc/modules. Cheers, Jaap ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Right timing for a queue call
Hello everybody, I'd like to have more detailed records for calls related to queues. For instance, if A enters in queue X, waits for Y secs and then talks to peer Z for T seconds, I'd like to have two entries in my CDR: - src: A, dst: X, duration: Y, state: ANSWERED - src: A, dst: Z, duration: T, state: ANSWERED This independently from how many peers the Queue app calls without success (peer not connected or not answering). The only way I could think of was giving an unique userfield to all the calls related to the call from A to the queue X, e.g.: - A - X usrfield: AX-uniqueid - call to peer B that doesn't answer, usrfield: AX-uniqueid - call to peer C that isn't available, usrfield: AX-uniqueid - call to peer Z that answers, usrfield: AX-uniqueid And then do some math based on duration and call state in order to get the info I need. Do you think that it's a good idea? How can it be implemented? I see that uniqueid changes for each call in the scenario that I described, so I'm a bit stuck. I'm using asterisk 1.2.2x (I know that I should migrate.. This is the last release of our product that uses 1.2). Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
On 15:38, Thu 03 Jan 08, Remco Barendse wrote:
On Thu, 3 Jan 2008, Benchev wrote:
Basically Grandstream HT286 is a single port FXS ATA.
In order to interconnect GSM gateway one would need FXO.
Are you sure it gives you new dialing tone or this is the * itself
you hear?
Yes, i am positive that i get a new dialtone from the GSM Gateway.
If i dial DTMF codes from a SIP phone connected to Asterisk, i can see the
digits appear in the display of the GSM Gateway. But it is a bit
incovenient to call an internal extension, wait for the dialtone and then
punch in all the numbers of the cell phone i need to call.
I would prefer Asterisk to decide where / how to route the call and send
the DTMF inband to the ATA device.
Thanks!!
You can use the D option with the Dial command.
Something like this should work:
exten = _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recorded calls skipping
Jay Moore wrote: Steve Totaro wrote: Jay Moore wrote: Greetings, List. I'm having a problem where my recorded calls are skipping every 4-5 seconds are so. I can hear the caller (or callee) just fine and then a second or so of silence followed by the person talking again. I'm saving my calls as .gsm files and it's worked fine for the past 11 months. I make sure I remove the recorded files from my Asterisk box and put them onto our fileserver, so it's not an issue of disk space. No other settings have been changed, so I'm not sure why my calls aren't being recorded properly now. Any thoughts? Thanks in advance, Jay You do not mention call volumes or simultaneous calls being recorded. If you are pushing around 70 or so simultaneous calls then you probably have an I/O issue with your hard drive. Although, I received complaints from the phone users about audio chopping before the recording were affected. I assume you are using the monitor app? Thanks, Steve Totaro At absolute maximum, we're probably recording 7-8 simultaneous calls, but most of the time it's 1-2. It's a newer rig, so I'm more inclined to think it's software and not hardware. Unfortunately, it's an older version of Asterisk, but I've had zero problems until the skipping calls, and if it ain't broke, don't fix it, right? :) I restarted Asterisk and it seems to have solved the problem -- for now at least. I'm a rookie when it comes to Asterisk, any suggestions on what to do to if it happens again? Thanks, Jay How much uptime was on the server? If and when it happens again, run top and look at CPU and memory usage. There have been buggy versions of Asterisk with memory leaks and such. It could be some other app or the OS as well. I would look at the version you are running and try to see what the latter version's release notes said as far as bug fixes. As you say, if it ain't broke, don't fix it. You may find a regular reboot is acceptable maintenance and it ain't broke. MRTG could be helpful as well as a weekly cron reboot (or whatever interval you feel comfortable with) Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lamps on Snom phones
Quoth Phil Knighton [EMAIL PROTECTED] I've incorporated the kind responses from other list members, such as setting call limits but to no avail! I've checked the function key settings on the Snom, and adjusted it to match the suggestion from another list member - nothing. Not one lamp on any of the phones will work, and I'm completely baffled as to why. A wild stab in the dark what version of the Snom firmware are you running? The lamps work for me on a Snom 370 running 7.1.28 and worked on other 7.1.last few releases but can't remember how far back. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recorded calls skipping
On Thu, Jan 03, 2008 at 12:22:57PM -0500, Steve Totaro wrote: How much uptime was on the server? If and when it happens again, run top and look at CPU and memory usage. Just the obvious comment here: In top I see: Mem:483588k total, 475448k used, 8140k free,64856k buffers Swap: 977896k total, 247624k used, 730272k free, 124500k cached whereas in the output of 'free' I see: total used free sharedbuffers cached Mem:483588 475300 8288 0 64868 124504 -/+ buffers/cache: 285928 197660 Swap: 977896 247624 730272 So the free number in top is generally meaningless. Add to it the size from buffers and cached to get a more realistic figure. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad Link on Website...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Not sure where to report this... http://www.asterisk.org/downloads Right hand download box, Asterisk 1.4.17 points to 1.4.1 Just a heads up. Stu - -- And all I can do is keep on telling you, I want you, I need you, But there aint no way Im ever gonna love you, Now dont be sad cause two out of three aint bad... -- Meatloaf - Two out of Three aint bad - Lyrics -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFHfRvkK69Y+xPZrWYRAt4UAJ9B5qZNoJmlmhpr7CApXBa+nQ3wAQCfU98g 5RRLuX3dA2h41bE4llqqhFo= =FUbn -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Link on Website...
Stuart Sheldon wrote: http://www.asterisk.org/downloads Right hand download box, Asterisk 1.4.17 points to 1.4.1 Sorry about that. It appears to be already fixed, though ... -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken
Quoth robert boardman [EMAIL PROTECTED]
Tzafrir Cohen wrote:
On Wed, Jan 02, 2008 at 07:41:40PM +, Russell Brown wrote:
Something in the, fairly, recent series of Asterisk updates has broken
DIGITAL call passthrough.
I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a
Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover
cable).
This PBX used to be able to make and receive DIGITAL type ISDN calls
through the Asterisk box... but something in the latest generation of
updates has broken it and although the calls seem to work the old PBX
just won't route traffic. Voice calls still work fine.
I've proven it's something in Asterisk by connecting the old PBX
directly to our ISDN PRI line and it still works fine.
What version is good? What version is bad?
Well I've had a fun few hours testing versions and eventually found out
what brings the problem to light.
I went back through versions of Asterisk et al until I got bored and
reinstated a complete backup from last August onto my Asterisk box
Voila! it worked (for inbound calls anyway).
Working my way forward in time... I eventually discounted all the
Asterisk, Zaptel and Libpri versions and boiled it down to me having
DYNAMIC_FEATURES=automon#autorecord#testfeature1
in the [globals] section of extensions.conf.
If I change this to DYNAMIC_FEATURES=automon then incoming DIGITAL calls
work. If DYNAMIC_FEATURES has anything more than this then it doesn't.
As a workaround, I've now got:
exten = _X.,n,Set(DYNAMIC_FEATURES=)
exten = _X.,n,Dial(Zap/g2/${EXTEN})
in the forward-to-my-old-PBX bit of the dialplan.
This works with 1.4.17, Zaptel 1.4.7.1 and libpri 1.4.3 (the current
stuff).
I have an outstanding problem with this,I have found that if you set
overlapdial to no on the internal leg ie connected to the pabx it works,
but you will have to set the pabx to dial en-block ie send all digits at
once
WRT Outgoing calls... this might help (unsetting DYNAMIC_FEATURES for
outbound stuff didn't do anything) but my old PBX is resisting dialing
en-block so calls fail :-(
Why-o-why setting DYNAMIC_FEATURES causes the PPP hookup from my old PBX
to fail I really can't imagine. Any developers care to comment? (I'm
happy to insert debug and send info)... or should I file a bug report?
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lamps on Snom phones
On Thu, 2008-01-03 at 18:00 +, Russell Brown wrote: Quoth Phil Knighton [EMAIL PROTECTED] I've incorporated the kind responses from other list members, such as setting call limits but to no avail! I've checked the function key settings on the Snom, and adjusted it to match the suggestion from another list member - nothing. Not one lamp on any of the phones will work, and I'm completely baffled as to why. A wild stab in the dark what version of the Snom firmware are you running? The lamps work for me on a Snom 370 running 7.1.28 and worked on other 7.1.last few releases but can't remember how far back. I had this same problem with a Snom 360 phone when we upgraded. We were running the 6.X firmware on the phones and could not get BLF to work on that phone. When we upgraded to the latest 7.X release everything began working again. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right timing for a queue call
Andrea Spadaccini wrote: Hello everybody, I'd like to have more detailed records for calls related to queues. For instance, if A enters in queue X, waits for Y secs and then talks to peer Z for T seconds, I'd like to have two entries in my CDR: - src: A, dst: X, duration: Y, state: ANSWERED - src: A, dst: Z, duration: T, state: ANSWERED This independently from how many peers the Queue app calls without success (peer not connected or not answering). The only way I could think of was giving an unique userfield to all the calls related to the call from A to the queue X, e.g.: - A - X usrfield: AX-uniqueid - call to peer B that doesn't answer, usrfield: AX-uniqueid - call to peer C that isn't available, usrfield: AX-uniqueid - call to peer Z that answers, usrfield: AX-uniqueid And then do some math based on duration and call state in order to get the info I need. Do you think that it's a good idea? How can it be implemented? I see that uniqueid changes for each call in the scenario that I described, so I'm a bit stuck. I'm using asterisk 1.2.2x (I know that I should migrate.. This is the last release of our product that uses 1.2). You can set some inheritable variable to uniqueid of channel before entering queue, and then in answer-macro (Dial(..,..,M()) set the CDR userfield to that variable. This would require use of Agent or Local channel, so you can do custom Dial for queue member. Works for me. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lamps on Snom phones
Watch the SIP Trace page on the Snom. 1. When it boots, it should send out a Subscribe message. 2. When the other phone is getting a call, the Snom should receive a Notify messages to tell the state. This might be a little out of date, but the main info is there: http://www.abptech.com/support/faqs/ | SP10 Regards, Shanon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Brown Sent: Thursday, January 03, 2008 12:01 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Lamps on Snom phones Quoth Phil Knighton [EMAIL PROTECTED] I've incorporated the kind responses from other list members, such as setting call limits but to no avail! I've checked the function key settings on the Snom, and adjusted it to match the suggestion from another list member - nothing. Not one lamp on any of the phones will work, and I'm completely baffled as to why. A wild stab in the dark what version of the Snom firmware are you running? The lamps work for me on a Snom 370 running 7.1.28 and worked on other 7.1.last few releases but can't remember how far back. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to retrieve my voice mail ... (password incorrect)
I used to work for Telefonica of Puerto Rico installing Asterisk, so I have installed few of them. I installed one last week (downloaded and installed the latest) and everything went beautiful and every thing works fine, however, my client has voice mail and no matter what phone I use, or what password I enter, or in which way I try I always get the same answer from the server: Password incorrect. I even deleted the extension and recreated it with a different number and get the same results. I looked in the voicemail.conf to verify that everything reflected correct in there and it is. The voicemail.conf shows ext. 200 with password 200 (which is what I enter). Any ideas? Thank you soo much for your help! William Herrera [EMAIL PROTECTED] William Herrera LAN/WAN Technical Consultant LAN Solutions http://www.lan-solutions.net/ www.lan-solutions.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A thougt
Is there any possibilletys to klick on a telephone nr an it will dail like the case in a mail program if you klick a url://a.b.se it opens a browser and in this case it would open a dailplane ?? Is there sucha thing ? Asking just out of curisoty /Fredrik Söderlund ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A thougt
I think Snapanumber might be what you are looking for. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Fredrik Söderlund Sent: Thursday, 3 January 2008 2:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] A thougt Is there any possibilletys to klick on a telephone nr an it will dail like the case in a mail program if you klick a url://a.b.se it opens a browser and in this case it would open a dailplane ?? Is there sucha thing ? Asking just out of curisoty /Fredrik Söderlund ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.17 - Breaks park announce?
Upgraded to 1.4.17 and found that the parking slot is not announced. Reverted back and all is well. Anyone else notice this behavior? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)
On Thursday 03 January 2008 22:15:07 William Herrera wrote: I installed one last week (downloaded and installed the latest) and everything went beautiful and every thing works fine, however, my client has voice mail and no matter what phone I use, or what password I enter, or in which way I try I always get the same answer from the server: Password incorrect. I even deleted the extension and recreated it with a different number and get the same results. I looked in the voicemail.conf to verify that everything reflected correct in there and it is. The voicemail.conf shows ext. 200 with password 200 (which is what I enter). Have you tried to change the dtmfmode of the respective peer to inband or rfc2833 or visa versa? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.17 - Breaks park announce?
Brent Torrenga wrote: Upgraded to 1.4.17 and found that the parking slot is not announced. Reverted back and all is well. Anyone else notice this behavior? If that is the case, put it on bugs.digium.com and it will get taken care of. I will try to take a look at it, as I think I may have changed that code recently. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - GEOPRIV and location based SIP services
Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is connected from office or home. Another option is to directly ask user or let them change default option but having this automatically detected is a bonus. Has anyone tried to build such location based SIP services ? I've read few lines about GEOPRIV which seems to be a building block for location based services but I could make sure if such DHCP extensions are implemented somewhere. Do you think GEOPRIV would help ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
On Thursday 03 January 2008 16:38:35 Remco Barendse wrote: On Thu, 3 Jan 2008, Benchev wrote: Basically Grandstream HT286 is a single port FXS ATA. In order to interconnect GSM gateway one would need FXO. Are you sure it gives you new dialing tone or this is the * itself you hear? Yes, i am positive that i get a new dialtone from the GSM Gateway. If i dial DTMF codes from a SIP phone connected to Asterisk, i can see the digits appear in the display of the GSM Gateway. But it is a bit incovenient to call an internal extension, wait for the dialtone and then punch in all the numbers of the cell phone i need to call. I would prefer Asterisk to decide where / how to route the call and send the DTMF inband to the ATA device. Yep. I've found a gsm gateway that does ...calls from VoIP to GSM and GSM to VoIP and uses the SIP protocols so is ideal for use as a SIP Trunk on many SIP based VoIP PBX Phone Systems... Sorry, didn't know such a thing exists. I don't think it matters dialing DTMF or not a simple dialplan trick should do. From home (Europe) I do: [gsm-out] exten = _0N.,1,Dial(SIP/gsm_gateway) exten = _0N.,2,Hangup Means all calls starting with zero and have digits from 2-9 afterwards go here. The mobile numbers start with 088 or 089. Otherwise I dial 01 for US and 011 for International. These are just ideas. You could figure out something else that fits your needs. Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF trouble
- Original Message - From: Lars Bensmann [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, January 03, 2008 4:24 PM Subject: Re: [asterisk-users] BLF trouble Does anybody have an idea where I can start looking to fix this? Or is this regular behaviour of asterisk that it does not show an extension as busy when it initiated the call? Thanks, Lars -- How do you have BLF set up ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to automaticaly closecallswhenAsterisk didn't receive the bye request ?
- Original Message - From: Steve Langstaff [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 03, 2008 11:49 AM Subject: Re: [asterisk-users] How to automaticaly closecallswhenAsterisk didn't receive the bye request ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B From: Jared Smith [EMAIL PROTECTED] There is a SIP timers patch in the bug tracker (see http://bugs.digium.com/view.php?id=10665) that currently implements this, and it's being tested in the team/group/sip_session_timers/ branch in SVN. Please test this out and help provide feedback, so that we can get this put into Asterisk in time for the next major release. Jared, I would think of using rtptimeout. There is a reason why you did not mention it and I am curious as to why. Does rtptimeout help if you are using canreinvite=yes ? Nope which Jared just explained to me. I am so used to not allowing invites that this one just went right over my head. Zoom. What was that ? ;) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)
The created extension its set to default (rfc2833). This is something I have never had the need to change ... (with the older versions of Asterisk) WH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benchev Sent: Thursday, January 03, 2008 4:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Unable to retrieve my voice mail ... (password incorrect) On Thursday 03 January 2008 22:15:07 William Herrera wrote: I installed one last week (downloaded and installed the latest) and everything went beautiful and every thing works fine, however, my client has voice mail and no matter what phone I use, or what password I enter, or in which way I try I always get the same answer from the server: Password incorrect. I even deleted the extension and recreated it with a different number and get the same results. I looked in the voicemail.conf to verify that everything reflected correct in there and it is. The voicemail.conf shows ext. 200 with password 200 (which is what I enter). Have you tried to change the dtmfmode of the respective peer to inband or rfc2833 or visa versa? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2759 (20080101) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com __ NOD32 2759 (20080101) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - GEOPRIV and location based SIP services
Can you provide more details on what you are trying to do. Your explanation is a bit confusing - sounds interesting but just want to make sure I have your idea right. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Thursday, 3 January 2008 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT - GEOPRIV and location based SIP services Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is connected from office or home. Another option is to directly ask user or let them change default option but having this automatically detected is a bonus. Has anyone tried to build such location based SIP services ? I've read few lines about GEOPRIV which seems to be a building block for location based services but I could make sure if such DHCP extensions are implemented somewhere. Do you think GEOPRIV would help ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to retrieve my voice mail ... (password incorrect)
Some of the grandstream phones refuse to listen to Asterisk, so you have to set them manuallygr. PaulH On Thu, 2008-01-03 at 17:23 -0400, William Herrera wrote: The created extension its set to default (rfc2833). This is something I have never had the need to change ... (with the older versions of Asterisk) WH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benchev Sent: Thursday, January 03, 2008 4:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Unable to retrieve my voice mail ... (password incorrect) On Thursday 03 January 2008 22:15:07 William Herrera wrote: I installed one last week (downloaded and installed the latest) and everything went beautiful and every thing works fine, however, my client has voice mail and no matter what phone I use, or what password I enter, or in which way I try I always get the same answer from the server: Password incorrect. I even deleted the extension and recreated it with a different number and get the same results. I looked in the voicemail.conf to verify that everything reflected correct in there and it is. The voicemail.conf shows ext. 200 with password 200 (which is what I enter). Have you tried to change the dtmfmode of the respective peer to inband or rfc2833 or visa versa? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2759 (20080101) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com __ NOD32 2759 (20080101) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Asterisk Appliance voicemail logs
That is correct; we would not recommend using just *any* CF card, as the write speed of the card needs to be pretty high to be able support multiple voicemail messages being written simultaneously. With that said, though, it is possible to use a higher capacity CF card, but my previous response that said it was 'easy' was a bit of an overstatement :-) It can be done, and our support department does know how to get you the files you would need to populate the replacement card. Kevin, In terms of ease, what is actually stored on the card, is it possible to simply place the card in a cf reader connected to a usb port on a linux box, tar up the contents and untar the contents on a larger cf card? Or is this something that would require a dd? Greg p.s. Sorry to put you over the coals... you can punch jarron b. for that one ;P No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1207 - Release Date: 1/2/2008 11:29 AM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Asterisk Appliance voicemail logs
Yea, sounds like they've planned for this issue. Kevin, is there an sdk that can be used to create our own binaries should we want to add modular support for something? Like say mysql cdr's? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murrell Sent: Monday, December 31, 2007 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium Asterisk Appliance voicemail logs On Mon, 2007-12-31 at 14:16 -0600, Kevin P. Fleming wrote: No. The files to repopulate the CF card are available to users who have active support subscriptions and they can replace the card. Users can also, of course, make a backup copy of the card on a new card when they receive the unit and have a ready-to-install replacement should any problems occur. That's all fair enough then. I was just concerned with the message that was being sent along with the replacing the CF card is unsupported message. b. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1207 - Release Date: 1/2/2008 11:29 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1207 - Release Date: 1/2/2008 11:29 AM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - GEOPRIV and location based SIP services
Olivier wrote: Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is connected from office or home. Another option is to directly ask user or let them change default option but having this automatically detected is a bonus. Has anyone tried to build such location based SIP services ? I've read few lines about GEOPRIV which seems to be a building block for location based services but I could make sure if such DHCP extensions are implemented somewhere. Do you think GEOPRIV would help ? Regards Hi Oliver, Linux Journal had an article about timezone handling in asterisk with perlscript for checking the GeoIP database with the IP adr. from the location db. Maybe that could give you a clue how to solve your question. http://www.linuxjournal.com/article/9190 The challange with GEOPRIV is that its rarely used so I would recomend GeoIP, http://www.maxmind.com. /Mats ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Right timing for a queue call
Ciao Atis,
Do you think that it's a good idea? How can it be implemented? I see that
uniqueid changes for each call in the scenario that I described, so I'm a
bit stuck.
I'm using asterisk 1.2.2x (I know that I should migrate.. This is the last
release of our product that uses 1.2).
You can set some inheritable variable to uniqueid of channel before
entering queue, and then in answer-macro (Dial(..,..,M()) set the CDR
userfield to that variable. This would require use of Agent or Local
channel, so you can do custom Dial for queue member. Works for me.
Thanks for your answer.
It didn't seem to work for me, as I already tried it.
Before calling the Queue app I set a variable to uniqueid, and then in the
context called from the Queue app I tried accessing it and it doesn't work.
I don't have the code right here (I'm at home), but it was something like
queues.conf
[somequeue]
member = Local/[EMAIL PROTECTED]
...
..
extensions.conf
[queue-caller]
exten = s,1,Set(TEST=a)
exten = s,n,Queue(various args)
[somecontext]
exten = 211,1,NoOP(${TEST})
exten = 211,n,Dial(SIP/211)
And then NoOp prints nothing.
Maybe I made some mistakes in the syntax (it's 2 AM here, and I'm just back
home), but I think that the concept is right.
Is this the same kind of setup that you have? Does it work for you?
Can you post some simplified working code?
Thanks in advance! :)
--
Dott. Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisc for Taking Calls for Radio
Hello Asterisc-Users List, I am new to the list. I joined with a question in mind: How would you set up an asterisc box so that: (A) Someone dials a number (B) They are presented with a menu (C) Entering a number, like 1, connects a call to me. (D) I am on a mixing board, running an internet radio show. I want to run asterisc into the board, and run an output from the board to asterisc. Is that possible using a soundcard? I don't really want to spend money. (E) I want the board to start wringing when I get a call, and I want the call audio to the board as well. I also would like it if I could not use my local phone line. I would prefer something like a free internet based number. The box will not need to be able to call out, so that's not a problem. A friend of mine uses asterisc, and has a free internet based number for asterisc. I would like to do the same. I hope this is possible, and thanks in advance. Shane -- -Shane Blog: http://blind-geek.com/blog/ CoOwner: http://sjtechzone.com AIM: inhaddict Skype: chatter8712 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - GEOPRIV and location based SIP services
MatsK wrote: Olivier wrote: Hi, I'm wondering whether or not it is achievable to build a web based click2dial application that could automatically detect that a user is connected from office or home. Another option is to directly ask user or let them change default option but having this automatically detected is a bonus. Has anyone tried to build such location based SIP services ? I've read few lines about GEOPRIV which seems to be a building block for location based services but I could make sure if such DHCP extensions are implemented somewhere. Do you think GEOPRIV would help ? Regards Hi Oliver, Linux Journal had an article about timezone handling in asterisk with perlscript for checking the GeoIP database with the IP adr. from the location db. Maybe that could give you a clue how to solve your question. http://www.linuxjournal.com/article/9190 The challange with GEOPRIV is that its rarely used so I would recomend GeoIP, http://www.maxmind.com. The problem with location tracking via IP is that, often, the entity who is the current owner of the IP at the time is not the same entity that has that IP space registered with ARIN. For example, I plugged in the serial side IP of one of the IP T1s in my office, and what comes up is my provider's corporate headquarters in MA, not my office in NJ. There are, however, other ways to do this. You could opt to have the user choose what location they're at and then drop a cookie on the user's browser at that location that will allow the browser to remember what location it's supposed to be. This isn't without its drawbacks either. There are many users that have cookies disabled because of privacy concerns, but at least with this approach the barrier to get it working again is generally in the hands of the user or their administrator. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automatic call marking an extension
hello list, happy new year to all, also to digium for their great work with asterisk . I want to make an automatic call marking an extension from my dial plan , an example that when marking the extension 100, tell me it records their message, mark the hour of their automatic call and at the end it marks the extension of the automatic call as I can make that, do they give me some ideas? greetings rickygm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.278 RTP conversation capture, please.
nothing? :'( On Dec 25, 2007 1:59 AM, Kerry S [EMAIL PROTECTED] wrote: Unfortunately I don't have a server set up that supports G.728. I'm asking for a software project. They also don't have the immediate resources. The goal of the project is to have a comprehensive VoIP conversation capture software suit for Windows. If you can procure one I would be most thankful. If you do not have a server set up for file sharing you could use http://rapidshare.com or something. On Dec 23, 2007 2:20 AM, Dovid B [EMAIL PROTECTED] wrote: Why don't you run tcpdump on any SIP server ? (Or are you emailing here because you don't have one and need one ? If that is the case can I ask why you need it ?) - Original Message - *From:* Kerry S [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Wednesday, December 19, 2007 2:11 AM *Subject:* [asterisk-users] G.278 RTP conversation capture, please. Hello all, I have a bit of a request. I need a wireshark capture of a SIP conversation using g.728. I don't need anything fancy, just a call and have both ends say hi to each other. hopefully someone out there can help me. Thank you all. This list has been of use many times in the past, even though I tend to stay quiet. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisc for Taking Calls for Radio
On Thursday 03 January 2008 19:36:10 Shane D wrote: Hello Asterisc-Users List, That's Asterisk, with a K. I do have to say, I've never seen that particular misspelling before, though. I am new to the list. I joined with a question in mind: How would you set up an asterisc box so that: (A) Someone dials a number (B) They are presented with a menu (C) Entering a number, like 1, connects a call to me. (D) I am on a mixing board, running an internet radio show. I want to run asterisc into the board, and run an output from the board to asterisc. Is that possible using a soundcard? I don't really want to spend money. Sure, just use chan_alsa (Console driver). (E) I want the board to start wringing when I get a call, and I want the call audio to the board as well. I also would like it if I could not use my local phone line. I would prefer something like a free internet based number. You're not going to get a free number connected to the PSTN, although you could publish a SIP address for no fee. For a PSTN number, you're going to need to pay at least a monthly fee; however, it will probably be low. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.278 RTP conversation capture, please.
Asterisk doesn't support g728. Any idea what does? PaulH On Thu, 2008-01-03 at 20:57 -0600, Kerry S wrote: nothing? :'( On Dec 25, 2007 1:59 AM, Kerry S [EMAIL PROTECTED] wrote: Unfortunately I don't have a server set up that supports G.728. I'm asking for a software project. They also don't have the immediate resources. The goal of the project is to have a comprehensive VoIP conversation capture software suit for Windows. If you can procure one I would be most thankful. If you do not have a server set up for file sharing you could use http://rapidshare.com or something. On Dec 23, 2007 2:20 AM, Dovid B [EMAIL PROTECTED] wrote: Why don't you run tcpdump on any SIP server ? (Or are you emailing here because you don't have one and need one ? If that is the case can I ask why you need it ?) - Original Message - From: Kerry S To: asterisk-users@lists.digium.com Sent: Wednesday, December 19, 2007 2:11 AM Subject: [asterisk-users] G.278 RTP conversation capture, please. Hello all, I have a bit of a request. I need a wireshark capture of a SIP conversation using g.728. I don't need anything fancy, just a call and have both ends say hi to each other. hopefully someone out there can help me. Thank you all. This list has been of use many times in the past, even though I tend to stay quiet. __ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registration from sip failed for ACL error (permit/deny)
Would this be a firewall problem? chan_sip.c handle_request_register: Registration from sip failed for ACL error (permit/deny) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration from sip failed for ACL error (permit/deny)
At 23:45 1/3/2008, Doug wrote: Would this be a firewall problem? chan_sip.c handle_request_register: Registration from sip failed for ACL error (permit/deny) Nope. I just needed to reload the configuration so that the phone could register. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A thougt
dean, fredrik, when I installed skype, ugh, it asked me if I wanted to link phone numbers on the web page to be click2dial... I did it and every phone number on a web page was a link... I ended up turning it off... it was too annoying... so there are some plug-ins out there that can do that sort of thing... daveC Dean Collins wrote: I think Snapanumber might be what you are looking for. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Fredrik Söderlund Sent: Thursday, 3 January 2008 2:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] A thougt Is there any possibilletys to klick on a telephone nr an it will dail like the case in a mail program if you klick a url://a.b.se it opens a browser and in this case it would open a dailplane ?? Is there sucha thing ? Asking just out of curisoty /Fredrik Söderlund ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agents and AddQueueMember
Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agents and AddQueueMember
Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use AddQueueMember to add the SIP [Local] channel to queue. The queue logs come like 1197953879|1197953876.34|Auth-Enq|Local/[EMAIL PROTECTED]|CONNECT|3|1197953876.35 Previously it used to come like 1197013076|1197013055.27|Auth-Enq|Agent/1001|CONNECT|21|1197013055.30 Here the problem is that there is no way to find number of calls taken by a person, because there is no agent abstraction here. What is the recommended way to work around this problem ? raj PS: Apologies for my previous mail, that was sent when I clicked the wrong button. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
