Re: [asterisk-users] Waiting for a call to complete with AMI Originate
Matt Riddell li...@venturevoip.com writes:
On 22/7/09 7:24 PM, Scott Gifford wrote:
[...]
In this case, I don't seem to have enough information to tell when the
call has failed and I should give up. I do get a Hangup event, but I
don't see a way to distinguish it from other hang-up events from other
calls.
For doing fax broadcasting we use the UserEvent function.
exten =
h,n,UserEvent(SmoothTorque|SmoothTorqueFax:${PHASEESTATUS}-${campaignid}-${phonenumber})
Then in the back end we parse the results.
Thanks, that worked great! I didn't know about the UserEvent app, it
will be a very useful trick to have up my sleeve. :-)
Scott.
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[asterisk-users] Using Of function SHARED
Dear All, i need help on Shared channel variable can any body have example of SHARED function which implemented in 1.6 version i can not find example regars Dhaval ___ -- 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] Friday 2009-07-24 12:00 EDT: Voxeo Labs on VoIP Users Conference
Hi all, You may have heard yesterday that Adhearsion and Voxeo have created a new baby, Voxeo Labs. From our (non-biz) point of view, I'd recommend following the blogs: http://blogs.voxeo.com/ to see how what they do might be of interest to you and your asterisk/voip activities, commercial or private. Since I myself know little about what this all means, I've invited a lot of bright people to our weekly conference. I've met Jay Phillips and Jason Goecke and they're interesting people to talk to on any subject, even outside the bounds of the usual geekdom. I only know Dan York from a few online exchanges and a visit he paid the VUC as a guest long ago. So I'm recommending you join us at 12 Noon EDT Friday July 24th to not only hear what all the buzz is about but also ask questions, make comments and drink the free virtual beer (you must be of legal virtual drinking age in your area). It's that virtual beer that got me in trouble on the second, non recorded, R-rated portion of our session last Friday. If you have a decent phone, it probably does g722 so join our call on the ZipDX wideband bridge: 200...@login.zipdx.com or call in to the Talkshoe g711 SIP URI: 7463#2262...@proxy.ideasip.com IRC back channel #voip-users-conference There's also a live stream and more information at http://VUC.me Thanks to Digium, OnSIP, e4strategies and ZipDX for all the help and support. Several regulars on the VUC come from those companies and provide a lot of insight in their areas. Should be a fun call this week. /r ___ -- 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] how to activate DND on 1.6.0.9
Hi, I want to activate DND on ast 1.6.0.9 with asterisk-gui. Is there special commands that i need to use during such script or simply writing a code in extensions.conf that checks if the user has a DND=yes value on ast. database and act according to that like forwarding call to voicemail or sending back a busy tone or playing a DND msg. And is there a way to notify a GPX_2000 for example for a DND status of another client?? ___ -- 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] astmanproxy?
Hmm I was given the impression that the .call files were risky due to locking issues... Is this no longer the case perhaps? I also require knowledge of whether the originate was successful or otherwise, with BUSY vs CONGESTION, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 22 July 2009 17:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] astmanproxy? When faced with this same problem, creating and FTPing .call files to the outgoing spool directory freed up the AMI for other functions. Plain, simple, and Just Worked I looked at, but never tried AstManProxy because I prefer to eliminate levels of complexity and points of failure rather than add, whenever possible. Not saying that this is your best solution, just what I found to be much more reliable than pounding the AMI. No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 8.5.392 / Virus Database: 270.13.24/2255 - Release Date: 07/22/09 18:00:00 ___ -- 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] Test Function if SIP Device is Still Alive
Hello! I am looking for a way to test if a SIP device is still alive or not. I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Thank you, Elliot ___ -- 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] CallerPres SIP headers Analog Phone
Ketema Harris wrote: hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file member:file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this? Have you set sendrpid = yes in your sip.conf? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] CallerPres SIP headers Analog Phone
Maybe https://issues.asterisk.org/view.php?id=71 should be re-opened because the north American vertical service codes are still hard- coded in Zaptel/Dahdi. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 and several clients behind NAT
Asterisk can 'ping' the clients behind NAT with the qualify-option so the NAT-tables and routes are kept open. What happens when one resets the router (where the NAT-tables are kept) ?? Do NAT-tables get flushed when a router is reset ?? Does the public IP-address needs to be a static IP-address ??? How can Asterisk use qualify to clients that are behind a dynamic public IP-address once registered ?? The clients are not aware that the public IP-address has changed and will not re-register automatically ?! Would dyndns be a solution ? Thanks for the feedback ! Jonas. On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote: jonas kellens wrote: Is it possible to have several clients behind NAT to register to an Asterisk-server with a public IP-address ? When Asterisk receives an incoming call, how will it know @ which private IP-address the client is reachable ? I guess it is impossible for Asterisk to directly contact the private client behind the NAT ?! Or to distinguish between the private clients ?! Is there an easy solution to this ? How does hosted IP-PBX services work then ?! Yes, this problem has a solution. The NAT gateway creates a UDP state mapping between internal source ports and external source (and destination, since most user agents are symmetrical nowadays) ports. The NAT gateway then allocates different external UDP ports for different connections being tracked in this manner. Consider, for example, two phones - 192.168.1.10 and 192.168.1.11 - registering to an outside SIP UAS through a NAT gateway whose public address is 67.194.23.55. The NAT gateway maps the source ports in a random or pseudorandom manner akin to: 192.168.1.10:5060 -- 67.194.23.55:32947 192.168.1.11:5060 -- 67.194.23.55:47948 If far-end NAT traversal is enabled on the UAS (in the case of Asterisk, that's nat=yes in sip.conf), the Contact URI supplied in the REGISTER message is ignored and the actual received IP and port on the network and transport layer is used in its place. The latter is what is stored as the contact binding. Later, a call comes in and the UAS maps it back to 67.194.23.55:47948 or 32947 depending on which registrant it is destined to go to. This scenario is not without its problems. Some user agents do not behave symmetrically. Some firewall/NAT router ALGs (application layer gateways) break this process, though they mean well and try to be helpful. But by far the most pressing problem is that many NAT gateways rather quickly age the temporary state information (internal:external UDP port mapping) out after a relatively short period of inactivity. That is why many far-end NAT traversal approaches implement a policy of periodically pinging the stored (received) contact with some sort of message that causes a bidirectional exchange of communication, and therefore causes the NAT gateway to reset its expiration timer for that connection state. In Asterisk, the OPTIONS messages generated when the qualify=yes option is enabled in sip.conf fulfill this function. Hope that helps, ___ -- 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] Test Function if SIP Device is Still Alive
Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 and several clients behind NAT
I've just had this Static/Dynamic IP issue in the last couple of days Any time the IP address changes the phone needs to be re-registered. This normally isn't that much of a problem as most people only reboot their routers about twice a year. However, here's a warning to anyone UK based. BT are now recycling their dynamic IPs on a nightly basis so if a customer has a SIP phone going through a BT dynamic IP service, they end up having to re-register on a daily basis. Best idea is to always use a static IP. Ish jonas kellens wrote: Asterisk can 'ping' the clients behind NAT with the qualify-option so the NAT-tables and routes are kept open. What happens when one resets the router (where the NAT-tables are kept) ?? Do NAT-tables get flushed when a router is reset ?? Does the public IP-address needs to be a static IP-address ??? How can Asterisk use qualify to clients that are behind a dynamic public IP-address once registered ?? The clients are not aware that the public IP-address has changed and will not re-register automatically ?! Would dyndns be a solution ? Thanks for the feedback ! Jonas. On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote: jonas kellens wrote: Is it possible to have several clients behind NAT to register to an Asterisk-server with a public IP-address ? When Asterisk receives an incoming call, how will it know @ which private IP-address the client is reachable ? I guess it is impossible for Asterisk to directly contact the private client behind the NAT ?! Or to distinguish between the private clients ?! Is there an easy solution to this ? How does hosted IP-PBX services work then ?! Yes, this problem has a solution. The NAT gateway creates a UDP state mapping between internal source ports and external source (and destination, since most user agents are symmetrical nowadays) ports. The NAT gateway then allocates different external UDP ports for different connections being tracked in this manner. Consider, for example, two phones - 192.168.1.10 and 192.168.1.11 - registering to an outside SIP UAS through a NAT gateway whose public address is 67.194.23.55. The NAT gateway maps the source ports in a random or pseudorandom manner akin to: 192.168.1.10:5060 -- 67.194.23.55:32947 192.168.1.11:5060 -- 67.194.23.55:47948 If far-end NAT traversal is enabled on the UAS (in the case of Asterisk, that's nat=yes in sip.conf), the Contact URI supplied in the REGISTER message is ignored and the actual received IP and port on the network and transport layer is used in its place. The latter is what is stored as the contact binding. Later, a call comes in and the UAS maps it back to 67.194.23.55:47948 or 32947 depending on which registrant it is destined to go to. This scenario is not without its problems. Some user agents do not behave symmetrically. Some firewall/NAT router ALGs (application layer gateways) break this process, though they mean well and try to be helpful. But by far the most pressing problem is that many NAT gateways rather quickly age the temporary state information (internal:external UDP port mapping) out after a relatively short period of inactivity. That is why many far-end NAT traversal approaches implement a policy of periodically pinging the stored (received) contact with some sort of message that causes a bidirectional exchange of communication, and therefore causes the NAT gateway to reset its expiration timer for that connection state. In Asterisk, the OPTIONS messages generated when the qualify=yes option is enabled in sip.conf fulfill this function. Hope that helps, ___ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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 - Do analog gateways detect a phone is plugged in or out ?
On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote: Olivier wrote: Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. Steve It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. A zero would indicate that nothing is attached, any other reading would indicate either equipment or at the least, show that the circuit is not open. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 - Do analog gateways detect a phone is plugged in or out ?
Is it usual for analog gateways to detect when an analog phone is plugged in or out ? It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. I read this with interest. The geek in me finds it amazing that they don't detect something plugged in. YOu think in the old days especially, it'd be easy based on what Steve says and that any proprietary system would do this to aid in setup and debugging, alarms etc. Nowadays, it might be a lot harder, although for SIP phones there are ways to detect any of the common ones. Druid does this during setup, for example. /r ___ -- 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] ExecIf and empty variables (early evaluation)
Benny Amorsen wrote:
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten = _X!,n,ExecIf($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
However, this doesn't actually work! Functions and variables on the
right hand side are evaluated BEFORE it is decided whether it needs to
be executed at all!
Try this, as I think the IF() function works differently (I could be wrong
though):
exten = _X!,n,Exec(${IF($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}:NoOp())})
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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[asterisk-users] Asterisk 1.4.25 and attended transfer
Hi all, I've a problem: I update my asterisk to version 1.4.25, and the attended transfer doesn't work. A call B, B press *2 and voice announce to digit internal and select internal of C. CORRECT A hear music on hold and B talks with C. CORRECT If B press *0, the call return to A. CORRECT if B hangup, .. also the call hangup Someone can help me??? Please! Marco ___ -- 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 - Do analog gateways detect a phone is plugged in or out ?
Steve Totaro schrieb: On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote: Olivier wrote: I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. Bonus point will be given for detecting the phone model and color as well. ;-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] odd behaviour with AGI and dial agent
Have you monitored the call from CLI with verbose set up? What happens if you use regular AGI instead of FastAGI? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Keiron Liddle Sent: Wednesday, July 22, 2009 10:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] odd behaviour with AGI and dial agent Hi, I have come across an odd problem. Basically I am transferring a call to an agent. The agent is logged in and set as paused. In order to find which agent to call I am using a fastagi script to just set a variable. When it falls through the agi script and dials the agent (using the variable) it doesn't connect the call properly to the agent. I get the beep but no audio (along with some other strange behaviour with the channel not hanging up properly, core show channels doesn't work properly). Now if I just set the variable in the dialplan (ie. no agi), or just hardcode the agent being called then it works fine. It seems that calling the fastagi is doing something to the channel which means that it doesn't work properly afterwards. I have also tried calling the agent in the agi with the same problems. Does anyone have any idea what the agi script could be doing to the channel/call, what it could be changing and how I can make it work properly. Keiron ___ -- 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] Asterisk 1.4.25 and attended transfer
Why didn't you just do 1.4.26 or 1.4 SVN? What release did you have? Did you (or the update) change your DTMF settings? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo Sent: Thursday, July 23, 2009 7:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.25 and attended transfer Hi all, I've a problem: I update my asterisk to version 1.4.25, and the attended transfer doesn't work. A call B, B press *2 and voice announce to digit internal and select internal of C. CORRECT A hear music on hold and B talks with C. CORRECT If B press *0, the call return to A. CORRECT if B hangup, .. also the call hangup Someone can help me??? Please! Marco ___ -- 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] Test Function if SIP Device is Still Alive
Hello Philipp, Thank you. I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could use? Regards, Elliot On Thu, Jul 23, 2009 at 12:47 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] CallerPres SIP headers Analog Phone
Yes. I have sendrpid = yes in sip.conf. CallerPres works fine with sip handsets. On Jul 23, 2009, at 4:29 AM, Ishfaq Malik wrote: Ketema Harris wrote: hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file member:file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this? Have you set sendrpid = yes in your sip.conf? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] Test Function if SIP Device is Still Alive
Elliot Murdock schrieb: I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could use? You could have a script execute asterisk -rx 'sip show peers' and read the status for each peer. On Thu, Jul 23, 2009 at 12:47 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Test Function if SIP Device is Still Alive
Hi You can retrieve it in real time using the AMI from a script http://www.voip-info.org/wiki/view/Asterisk+manager+API Ish Elliot Murdock wrote: Hello Philipp, Thank you. I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could use? Regards, Elliot On Thu, Jul 23, 2009 at 12:47 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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 - Do analog gateways detect a phone is plugged in or out ?
On Thu, Jul 23, 2009 at 7:50 AM, randulo spamsucks2...@gmail.com wrote: Is it usual for analog gateways to detect when an analog phone is plugged in or out ? It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. I read this with interest. The geek in me finds it amazing that they don't detect something plugged in. YOu think in the old days especially, it'd be easy based on what Steve says and that any proprietary system would do this to aid in setup and debugging, alarms etc. Nowadays, it might be a lot harder, although for SIP phones there are ways to detect any of the common ones. Druid does this during setup, for example. /r It has been such a long time but I seem to remember the Definity G3 (maybe others) were aware of the digital sets that were attached. On the other hand, I vaguely remember specifying the set in the UI, so I am not positive because I inherited that six cabinet beast of a beast. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] H323 situation
Hi, Still I can manage to have good incoming calls from h323. Can someone give me a hand? Regards, LS Date: Thu, 16 Jul 2009 15:46:43 +0100 From: Luis Silva luis.si...@dreamware.pt Subject: [asterisk-users] H323 situation To: asterisk-users@lists.digium.com Message-ID: 00ab01ca0624$3c9f69b0$b5de3d...@silva@dreamware.pt Content-Type: text/plain; charset=us-ascii Hi all, I have this installation: Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. I have a problem that is, when a call comes from H323 and goes to a Sip phone the asterisk sends two rtp streams to the sip. I checked this with tcpdump, save the payload (voice is in G711u), one is the ringing indication and the other is the voice coming from the user in h323 side. And worst they go to the same port. This causes that in the sip phone there are problems, when the call is answered sometimes we get the riging indication, others a mix of the two with very bad sound quality and others(few) a god audio call. The outgoing calls from sip to H323 are ok. I also tested an incoming call from a dahdi channel and from here everything is ok, only one rtp stream and a good call. By the way I had other problem that I fixed, but don't know if it was in the best way. The h323 box is a Cisco AS5300 (or 5350?) and when I was making outgoing calls the AS disconnected all of them after 10 sec. I investigated I noticed that the AS as a limitation to the G711 payload to 20 ms, and asterisk was using 150 ms. I resolve this changing in frame.c the codec value and recompile asterisk. There is simpler way to do this? Like changing values in codec.conf?... Regards LS ___ -- 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 - Do analog gateways detect a phone is plugged in or out ?
Philipp Kempgen wrote: Steve Totaro schrieb: On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote: Olivier wrote: I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. Bonus point will be given for detecting the phone model and color as well. ;-) Philipp Kempgen Yes, it's technically possible for the phone company to determine if there is a set or something connected to a phone line. It involved hitting the line with +130v dc test voltage and reversing it quickly and seeing how much capacatance kick there is. This kind of testing is normal for telco CO lines. FXS chan units or gateways normally do NOT have this built into them. The only exception I know about is SLC(Subscriber Line Concentrator, which is a generiac term for fiber or digital lines feed to telco boxes in the field). And even there the process was to have a cut-in relay and connect the out cable pair back to the CO via a dedicated copper pair to do these tests via a device called a PairGain Test Controller. I know because I was an 'expert' on them and traveled around going from telco CO to CO fixing them. In other words, there is some circuitry involved in doing these tests and I don't see any PBX, FXS chan unit or gateway manufacturer rushing to add more to this to their product line. They have not done it yet and I don't see anyone other than the phone company willing to spend the money to make it happen. To keep this on topic for Philipp's remark, the only bonus points we assigned was to correctly guess how many phones were attached to the phone lineGRIN! Lyle Giese LCR Computer Services, 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] Music on hold based on user
Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 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
Re: [asterisk-users] Music on hold based on user
On Thu, Jul 23, 2009 at 9:35 AM, Juan C. Crespo R.jcre...@ifxnw.com.ve wrote: Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 Seems like a perfect use of SetMusicOnHold.. http://www.voip-info.org/wiki/view/Asterisk+cmd+SetMusicOnHold -jonathan ___ -- 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] Analog FXO or IAX DIDS for new facility?
I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read Asterisk: The Future of Telephony and done a fair amount of googling. I am completely sold on Asterisk, and the new building's phones will be a mix of SIP handsets and softphones. I am confused about one thing: Should we be getting a block of analog circuits from the local telco (probably ATT), connected to the server's FXO cards for in-bound and out-bound POTS calls; or should we get a block of DIDS numbers from one of the plethora of providers available over the Internet, and then have our server connect POTS calls by IAX to the DIDS provider? We are unsure whether we are going to have separate numbers for everyone in the organization, or just 1 US phone number, with everyone in the org having their own extension number. That probably largely depends upon cost. We will have 75 people in the building. We have no data on call patterns or usage (because our legacy system belongs to our current facilities host), but we currently have 4 lines for 35 people and on unusual occasions they all get busy. An additional consideration is that we also have 300 other people scattered literally world-wide, and the next logical future step is to start providing VOIP links for them, as well. Thanks in advance for your advice. Any other suggestions, such as # of lines sizing info or reputable DIDS vendors (if that's the answer) are also appreciated. -- Sincerely Yours, Stephen P. Fierbaughstep...@fierbaugh.org Pioneer Bible Translatorsstephen.fierba...@pbti.org Pronounced: Fire as in hot, Bah as in humbug! John 3:16 in over 3,000 languages. ___ -- 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] Analog FXO or IAX DIDS for new facility?
My .02 - IAX may not be an option and is probably not a good one if it is. It requires a good bit of overhead to work reliably and well. You won't go wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16 port card and make sure you get the card away from any existing IRQ's, especially the RAID one. If you went SIP DID instead of FXO, this would make putting your world-wide folks in an easier task. IMO a pretty good rule-of-thumb is that a line for every 8 folks will generally work pretty well, with a minimum of 3 lines. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Fierbaugh (PBT) Sent: Thursday, July 23, 2009 9:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Analog FXO or IAX DIDS for new facility? I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read Asterisk: The Future of Telephony and done a fair amount of googling. I am completely sold on Asterisk, and the new building's phones will be a mix of SIP handsets and softphones. I am confused about one thing: Should we be getting a block of analog circuits from the local telco (probably ATT), connected to the server's FXO cards for in-bound and out-bound POTS calls; or should we get a block of DIDS numbers from one of the plethora of providers available over the Internet, and then have our server connect POTS calls by IAX to the DIDS provider? We are unsure whether we are going to have separate numbers for everyone in the organization, or just 1 US phone number, with everyone in the org having their own extension number. That probably largely depends upon cost. We will have 75 people in the building. We have no data on call patterns or usage (because our legacy system belongs to our current facilities host), but we currently have 4 lines for 35 people and on unusual occasions they all get busy. An additional consideration is that we also have 300 other people scattered literally world-wide, and the next logical future step is to start providing VOIP links for them, as well. Thanks in advance for your advice. Any other suggestions, such as # of lines sizing info or reputable DIDS vendors (if that's the answer) are also appreciated. -- Sincerely Yours, Stephen P. Fierbaughstep...@fierbaugh.org Pioneer Bible Translatorsstephen.fierba...@pbti.org Pronounced: Fire as in hot, Bah as in humbug! John 3:16 in over 3,000 languages. ___ -- 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] Music on hold based on user
Juan C. Crespo R. schrieb:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play music
1, if another from Technical Support put on hold the call play music 3,
something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
Some dialplan logic around Set(CHANNEL(musicclass)=...) should do
the trick I guess.
Maybe the easiest way (in Asterisk 1.6) would be to add
setvar=musicclass=admin / setvar=musicclass=support / ...
to your SIP peers and then do something like
_X. = {
Set(CHANNEL(musicclass)=${SIPPEER(${EXTEN},chanvar[musicclass])})
Dial(SIP/${EXTEN});
}
in your dialplan (untested).
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] Music on hold based on user
Depending on how your dialplan is set you can use the SetMusicOnHold application after creating classes in your musiconhold.conf http://www.asteriskguru.com/tutorials/setmusiconhold.html Ish Juan C. Crespo R. wrote: Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] Analog FXO or IAX DIDS for new facility?
On Thu, Jul 23, 2009 at 5:08 PM, Danny Nicholasda...@debsinc.com wrote: My .02 - IAX may not be an option and is probably not a good one if it is. It requires a good bit of overhead to work reliably and well. You won't go wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16 port snip I second what Danny said, go for SIP DID, there are many good providers and you could even have local DID in different countires if that made it easier for your correspondents. There are IAX providers too , though if you have a compelling reason to use IAX. Go with a solid, long running company on the DIDs. /r ___ -- 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 - Do analog gateways detect a phone is plugged in or out ?
Lyle Giese wrote: Philipp Kempgen wrote: Steve Totaro schrieb: On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote: Olivier wrote: I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. Bonus point will be given for detecting the phone model and color as well. ;-) Philipp Kempgen Yes, it's technically possible for the phone company to determine if there is a set or something connected to a phone line. It involved hitting the line with +130v dc test voltage and reversing it quickly and seeing how much capacatance kick there is. This kind of testing is normal for telco CO lines. FXS chan units or gateways normally do NOT have this built into them. The only exception I know about is SLC(Subscriber Line Concentrator, which is a generiac term for fiber or digital lines feed to telco boxes in the field). And even there the process was to have a cut-in relay and connect the out cable pair back to the CO via a dedicated copper pair to do these tests via a device called a PairGain Test Controller. I know because I was an 'expert' on them and traveled around going from telco CO to CO fixing them. In other words, there is some circuitry involved in doing these tests and I don't see any PBX, FXS chan unit or gateway manufacturer rushing to add more to this to their product line. They have not done it yet and I don't see anyone other than the phone company willing to spend the money to make it happen. To keep this on topic for Philipp's remark, the only bonus points we assigned was to correctly guess how many phones were attached to the phone lineGRIN! Sane people just make a call, and see if the darned thing works. :-) 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
[asterisk-users] detect keys before agi starts
I am running an AGI 1.4.26 A person answers the call, and presses a DIGIT really fast. Perhaps while the AGI is still starting up. Is there anyway to get that digit? When doing wait for digit if my AGI is up and running I seem to get the digit every time. Is there a way/method to get ANY digits pressed while during startup? Thanks, Jerry ___ -- 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] x-lite settings to reach asterisk
Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet -DSL transparent bridge -router -asterisk -softphone x-lite attempts to login and register, but times out. There must be some setting I'm missing. Any help appreciated. Tom ___ -- 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] dinamic queue distribution
Hello I have 2 queues and I would like to send calls to queue_1 and queue_2 dynamically. For example: If I have 10 agents logged (2 in queue_1 and 8 in queue_2) I want 20% of the calls to be sent to queue_1 and 80% to queue_2 Is this possible? Is there a way I can see how many logged (or available) agents I have in a queue before sending a call? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] x-lite settings to reach asterisk
Does your asterisk has a private or public IP? Is your X-Lite in your LAN or outside? If X-Lite is outside the Lan, you need to forward all traffic coming to your Lan in port 5060, to asterisks private IP. Activate SIP debug in asterisk CLI to check if the traffic is getting to asterisk. Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt Tom Poe wrote: Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet -DSL transparent bridge -router -asterisk -softphone x-lite attempts to login and register, but times out. There must be some setting I'm missing. Any help appreciated. Tom ___ -- 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 -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] x-lite settings to reach asterisk [SOLVED]
Joao Gomes Pereira wrote: Does your asterisk has a private or public IP? Is your X-Lite in your LAN or outside? If X-Lite is outside the Lan, you need to forward all traffic coming to your Lan in port 5060, to asterisks private IP. Activate SIP debug in asterisk CLI to check if the traffic is getting to asterisk. Joao Pereira Joao: Thanks for responding. Last night, I gave up trying to get the settings right. Then, today, right after posting, I brought up x-lite to give it another go, and the dumb thing logged in successfully and all tests for in-house connections are working. Not sure what happened, but this step is now completed. I'll now start in to see if I can set things up to call out. I have the pre-configured PBX in a Flash computer. I'm following the book by Blanchas. Wish me luck. Tom ___ -- 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] PRI call progress issue
I've got a couple of PRIs. When I call out on them from internal SIP phones, I will get ringing if the dialed number is ringing, but if the dialed number is busy I'll get dead air. Can anyone suggest ways to trouble shoot this? Don't seem to having any other problems with the PRIs. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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] ExecIf and empty variables (early evaluation)
On Thursday 23 July 2009 07:24:46 Leif Madsen wrote:
Benny Amorsen wrote:
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten = _X!,n,ExecIf($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
However, this doesn't actually work! Functions and variables on the
right hand side are evaluated BEFORE it is decided whether it needs to
be executed at all!
Try this, as I think the IF() function works differently (I could be wrong
though):
exten = _X!,n,Exec(${IF($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}:NoOp())})
You're incorrect. The same problem follows this one. Separating the
evaluation out into multiple steps is the only way I know to make this work
as expected:
exten = _X!,n,GosubIf(${LEN(${QueueName})}?waitingcount(foo,${QueueName}))
...
exten = _X!,1000
(waitingcount),Set(${${ARG1}}=${QUEUE_WAITING_COUNT(${ARG2})})
exten = _X!,n,Return
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with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies)
and Harry, BB, George (dogs)
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Re: [asterisk-users] dinamic queue distribution
You could do an AGI to get the queue information via AMI queue status, then
return variables to the dialplan and select the queue dynamically based on
that information.
[global]
CALLCOUNT=0
- exten = s,1,answer
- exten = s,2,AGI(questat.agi)
- exten = s,3,set(GLOBAL(CALLCOUNT)=[1 + ${CALLCOUNT}]))
- exten = s,4(check),Gotoif($[${QUEUE1} ${CALLCOUNT}]?queue2)
- exten = s,5,queue(1)
- exten = s,6,hangup
- exten = s,7(queue2),queue(2)
- exten = s,8,hangup
- exten = s,9(reset),Set(GLOBAL(CALLCOUNT)=0)
- exten = s,10,goto(default|s|check)
This needs some cleanup, but hopefully conveys the general idea.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes
Pereira
Sent: Thursday, July 23, 2009 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] dinamic queue distribution
Hello
I have 2 queues and I would like to send calls to queue_1 and queue_2
dynamically.
For example:
If I have 10 agents logged (2 in queue_1 and 8 in queue_2)
I want 20% of the calls to be sent to queue_1 and 80% to queue_2
Is this possible?
Is there a way I can see how many logged (or available) agents I have in
a queue before sending a call?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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Re: [asterisk-users] PRI call progress issue
David Ruggles wrote: is busy I'll get dead air. Can anyone suggest ways to trouble shoot this? Don't seem to having any other problems with the PRIs. It'd be nice to start with what version of Asterisk, what distro, who is your service provider and snippets of your config. On our PRIs we show busy when a line is busy. My systems are out of Michigan and Indiana. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] PRI call progress issue
Apologies. Didn't mean to omit key information, I doubt it's a problem with * because everything else is working great so I was asking for help on troubleshooting the PRI. Anyway, here's the 411: Asterisk 1.4.20, CentOS 5.2 Service Providers: Quest Deltacom, Local Loops provided by Embarq What snippets of the config would be helpful? I've included zapata.conf and zaptel.conf below. zapata.conf: ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:8 bus:1 span:5] wanpipe5 switchtype=dms100 context=from-pstn group=0 signalling=pri_cpe channel =97-119 ;Sangoma A102 port 2 [slot:8 bus:1 span:6] wanpipe6 switchtype=dms100 context=from-pstn group=1 signalling=pri_cpe channel =121-143 zaptel.conf: # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:8 bus:1 span:5] wanpipe5 span=5,0,0,esf,b8zs bchan=97-119 hardhdlc=120 #Sangoma A102 port 2 [slot:8 bus:1 span:6] wanpipe6 span=6,0,0,esf,b8zs bchan=121-143 hardhdlc=144 Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, July 23, 2009 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI call progress issue David Ruggles wrote: is busy I'll get dead air. Can anyone suggest ways to trouble shoot this? Don't seem to having any other problems with the PRIs. It'd be nice to start with what version of Asterisk, what distro, who is your service provider and snippets of your config. On our PRIs we show busy when a line is busy. My systems are out of Michigan and Indiana. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] PRI call progress issue
David Ruggles wrote: [channels] I also have listed pridialplan=unknown immediate=no ;Sangoma A102 port 1 [slot:8 bus:1 span:5] wanpipe5 I've got two sites running the Sangoma A101, what version of your wanpipe drivers are you running (Mine are probably very outdated WANPIPE Release: 3.1.4) switchtype=dms100 I've read that the provider has to support this specifically, if I recall correctly, it allows for redirecting of a call and frees up the line? Mine is set to national Just an FYI, I found in the archive, while searching for dms100, that Jared Smith stated that you'll also need to set facilityenable=yes in the zaptap.conf hardhdlc=120 Mine is set to dchan= But, this is probably because of the age of my drivers. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 Of function SHARED
On Thursday 23 July 2009 02:05:38 DHAVAL INDRODIYA wrote:
Dear All,
i need help on Shared channel variable
can any body have example of SHARED function which implemented in 1.6
version
It's actually fairly simple. On each channel, there is a space accessible for
other channels to write:
Set(SHARED(foo,SIP/123)=456)
or retrieve:
${SHARED(foo,SIP/123)}
The primary reason for having this space is writing out to another channel,
since you can already import variables (and functions) from another channel,
with the IMPORT function:
${IMPORT(SIP/123,CALLERID(name))}
Just remember that this is a _special_ variable space and not the main
variable space, so that other channels cannot mess with your execution
except when you explicitly want them to be able to do so.
--
Tilghman Teryl
with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies)
and Harry, BB, George (dogs)
___
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[asterisk-users] nortel cs 1000 swtich
Anyone successully connected to nortel cs 1000 switch? Care to share you switch settings? I have asterisk 1.4.25, libpri 1.4.7, dahdi We tried national and the verizon guy said that wasnt working... We tried 5ess and we can get external calls - but internal calls we have no audio. I see frame drops in the log file. I am setup for digium clocking - they are setup to receive clocking. We are using a single T1 card. Thanks, Jerry ___ -- 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] Analog FXO or IAX DIDS for new facility?
In australia, I would usually suggest a mix of E1 and SIP for calls - it doesn't cost any money to receive calls via E1, and redundancy is an old, valuable friend of mine. PaulH Stephen Fierbaugh (PBT) wrote: I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read Asterisk: The Future of Telephony and done a fair amount of googling. I am completely sold on Asterisk, and the new building's phones will be a mix of SIP handsets and softphones. I am confused about one thing: Should we be getting a block of analog circuits from the local telco (probably ATT), connected to the server's FXO cards for in-bound and out-bound POTS calls; or should we get a block of DIDS numbers from one of the plethora of providers available over the Internet, and then have our server connect POTS calls by IAX to the DIDS provider? We are unsure whether we are going to have separate numbers for everyone in the organization, or just 1 US phone number, with everyone in the org having their own extension number. That probably largely depends upon cost. We will have 75 people in the building. We have no data on call patterns or usage (because our legacy system belongs to our current facilities host), but we currently have 4 lines for 35 people and on unusual occasions they all get busy. An additional consideration is that we also have 300 other people scattered literally world-wide, and the next logical future step is to start providing VOIP links for them, as well. Thanks in advance for your advice. Any other suggestions, such as # of lines sizing info or reputable DIDS vendors (if that's the answer) are also appreciated. ___ -- 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 asterisk on a shared line
Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is not running, I can answer an incoming call from that phone. If asterisk is running, I can answer the call from a sip extension. Can I have it both ways? Can Asterisk back off if the secondary phone answers the call? Currently, if a call comes in and I answer it from the secondary phone Asterisk will continue to ring the sip extension and eventually drop into voicemail. ___ -- 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 asterisk on a shared line
On Thu, Jul 23, 2009 at 8:34 PM, Trevor Hammonds tre...@concipient.netwrote: Bill Lovett wrote: Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is not running, I can answer an incoming call from that phone. If asterisk is running, I can answer the call from a sip extension. Can I have it both ways? Can Asterisk back off if the secondary phone answers the call? Currently, if a call comes in and I answer it from the secondary phone Asterisk will continue to ring the sip extension and eventually drop into voicemail. Asterisk is a PBX, not an answering machine, so I would advise against this. It would be best to have Asterisk handle the phone line exclusively, 24/7. However, with that said, it is possible to accomplish what you are asking. Placing a telephone privacy/exclusion adapter on the line cord into Asterisk will cut off the phone line whenever a parallel telephone on the same line is picked up. This means that the instant you pick up any other phone on the line, it would cut off the line to Asterisk. Radio Shack used to sell a couple varieties of these. One was a two-way adapter with one side for phone and the other answering machine. You do not need to plug anything into the phone side for the device to work. The second device was just an inline exclusion device. I was unable to find these at Radio Shack's website. However, I found something similar at the following URLs: (See SER2A, SER2D, and SER3P at Sandman.com) http://www.sandman.com/lineshar.html http://www.trianglecables.com/telanmacorph.html http://www.iec-usa.com/cgi-bin/iec/COM9928 http://www.iec-usa.com/cgi-bin/iec/COM0006 Good luck! Sincerely, Trevor Hammonds Thanks, filed away in memory bank. I will probably order one very soon. Not for Asterisk of course but for annoying answering machines that pick up if you don't grab the phone in time and record your whole darn conversation. Get close and the feedback is a killer. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 1.4.25 and attended transfer
Marco Sambo a écrit : Hi all, I've a problem: I update my asterisk to version 1.4.25, and the attended transfer doesn't work. [...] Marco, attented transfer are broken in 1.4.25, please upgrade to 1.4.26 (see changelog). -- Daniel ___ -- 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] nortel cs 1000 swtich
Jerry Geis wrote: Anyone successully connected to nortel cs 1000 switch? Care to share you switch settings? I have asterisk 1.4.25, libpri 1.4.7, dahdi We tried national and the verizon guy said that wasnt working... We tried 5ess and we can get external calls - but internal calls we have no audio. I see frame drops in the log file. I am setup for digium clocking - they are setup to receive clocking. We are using a single T1 card. I have my Nortel Option 61C running CS1000 4.50 tied to Asterisk 1.4.21.2 libpri version 1.4.8 working pretty well. I had originally configured using 5ess and it worked except for CLID passing only one way which may or may not have been solved by the switch to QSIG because what I needed after QSIG was facilityenable = yes, but I digress. Below are all the config info for my system. On the Nortel side, my Tie line is on loop 3 and the route is also 3 as I like to keep it simple ;-) The D-Channel is on Channel 24. The D-Channel number is 7 and controlled by the MSDL card #8 port 2. The Interface(IFC) on DCH 7 is ISGF for QSIG. It also has to be set on the route (also IFC ISGF) I hope this helps. Dale zaptel.conf: * # Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 # Global data loadzone= us defaultzone = us ** Zapata.conf ** [trunkgroups] [channels] language=en context=from-internal signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ; Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) group=0,11 context=from-internal switchtype=qsig signalling=pri_cpe facilityenable = yes channel = 1-23 On the nortel side of the house ** ADAN for D-Channel 7 (D-Channel for PRI Tie to Asterisk) *** ADAN DCH 7 CTYP MSDL DNUM 8 PORT 2 DES ASTERISK USR PRI DCHL 3 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ISGF PINX_CUST 0 ISDN_MCNT 300 CLID OPT0 CO_TYPE STD SIDE NET CNEG 1 RLS ID 1 RCAP COLP NDI CCBI CCNI PRI DV3I CTI QMWI PR_TRIGS DIV 2 3 CNG 2 3 PR_RTN NO MBGA NO OVLR NO OVLS NO T310 120 T200 3 T203 10 N200 3 N201 260 K7 *** CEQU for Loop 3 (T-1 to Asterisk) *** DLOP NUM DCH FRM TMDI LCMT YALM TRSH 003 24 ESF NO B8S FDL 00 *** Route Data Block for Route 3 (Tie to Asterisk) *** TYPE RDB CUST 00 ROUT 3 DES TIE ASTERISK TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT VTRK NO NODE DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ISGF SBN NO PNI 0 NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC NO ISAR NO CPFXS YES DAPC NO INTC NO DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 8903 TCPP NO TARG 01 CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 ARDN NO AACR NO ** The TNB for a single trunk member(channel) in Route 3 ** DES ASTERISK TN 003 01 TYPE TIE CDEN SD CUST 0 TRK PRI PDCA 1 PCML MU NCOS 6 RTMB 3 1 B-CHANNEL SIGNALING TGAR 1 AST NO IAPG 0 CLS CTD DIP WTA LPR APN THFD HKD P10 VNL TKID AACR NO DATE 4 JUN 2009 No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 8.5.392 / Virus Database: 270.13.25/2256 - Release Date: 07/23/09 06:02:00 ___ -- 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 asterisk on a shared line
Short answer - no. Leave the box on 24/7, and run the POTS phone through an ATA, or another SIP phone. If power consumption and wear and tear is a consideration, use AstLinux on a thin client, and reduce your power consumption to under 30 Watts. John Novack Bill Lovett wrote: Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is not running, I can answer an incoming call from that phone. If asterisk is running, I can answer the call from a sip extension. Can I have it both ways? Can Asterisk back off if the secondary phone answers the call? Currently, if a call comes in and I answer it from the secondary phone Asterisk will continue to ring the sip extension and eventually drop into voicemail. ___ -- 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 -- Dog is my co-pilot ___ -- 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 asterisk on a shared line
An exclusion adapter is overkill. My Asterisk line card is the $10 Win modem card that I got from ebay. When you call my copper line, two devices see the inbound ringer: 1. The Uniden 5.8Ghz cordless phone base station that answers 95% of the calls 2. Asterisk with a win modem line card that: a. runs a perl AGI script to parse caller-id name and number b. rings a sip extension or c. answers the call and plays funny messages and DTMF tones at the telemarketers. Just make sure that Asterisk only RINGS the sip extensions but never sends the call to play a message or voicemail or any other Asterisk feature that will issue an implicit Answer and take the call. ___ -- 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 asterisk on a shared line
Shorter answer is yes :-). This is exactly how mine runs. The secret is that the copper interface will ring a SIP extension but just exit from the dialplan on noanswer. [main-copper] exten = s,1,Dial(SIP/22,69) and then nothing in my case. Generally my wife answers using a cordless phone set that is sharing the copper line with my Asterisk line card. The other benefit is that I actually parse caller-id name and number and optionally have Asterisk answer and torture telemarketers if there is a match. Otherwise it just rings my SIP extensions and will not seize the line unless I pickup extension 22. ___ -- 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 asterisk on a shared line
Bill Lovett wrote: Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is not running, I can answer an incoming call from that phone. If asterisk is running, I can answer the call from a sip extension. Can I have it both ways? Can Asterisk back off if the secondary phone answers the call? Currently, if a call comes in and I answer it from the secondary phone Asterisk will continue to ring the sip extension and eventually drop into voicemail. Asterisk is a PBX, not an answering machine, so I would advise against this. It would be best to have Asterisk handle the phone line exclusively, 24/7. However, with that said, it is possible to accomplish what you are asking. Placing a telephone privacy/exclusion adapter on the line cord into Asterisk will cut off the phone line whenever a parallel telephone on the same line is picked up. This means that the instant you pick up any other phone on the line, it would cut off the line to Asterisk. Radio Shack used to sell a couple varieties of these. One was a two-way adapter with one side for phone and the other answering machine. You do not need to plug anything into the phone side for the device to work. The second device was just an inline exclusion device. I was unable to find these at Radio Shack's website. However, I found something similar at the following URLs: (See SER2A, SER2D, and SER3P at Sandman.com) http://www.sandman.com/lineshar.html http://www.trianglecables.com/telanmacorph.html http://www.iec-usa.com/cgi-bin/iec/COM9928 http://www.iec-usa.com/cgi-bin/iec/COM0006 Good luck! Sincerely, Trevor Hammonds ___ -- 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 asterisk on a shared line
I get how everything is connected with your setup, but if you pick up the cordless phone to answer a call does the sip extension just keep ringing until it times out? I like the exclusion adapter idea because it sounds like it would let me keep my dialplan intact. But I do take John and Trevor's point about putting everything through asterisk and running it 24/7. It would make things a lot simpler. On Jul 23, 2009, at 8:58 PM, Tom Browning wrote: An exclusion adapter is overkill. My Asterisk line card is the $10 Win modem card that I got from ebay. When you call my copper line, two devices see the inbound ringer: 1. The Uniden 5.8Ghz cordless phone base station that answers 95% of the calls 2. Asterisk with a win modem line card that: a. runs a perl AGI script to parse caller-id name and number b. rings a sip extension or c. answers the call and plays funny messages and DTMF tones at the telemarketers. Just make sure that Asterisk only RINGS the sip extensions but never sends the call to play a message or voicemail or any other Asterisk feature that will issue an implicit Answer and take the call. ___ -- 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 asterisk on a shared line
On Thu, Jul 23, 2009 at 8:58 PM, Tom Browning ttbrown...@gmail.com wrote: An exclusion adapter is overkill. My Asterisk line card is the $10 Win modem card that I got from ebay. When you call my copper line, two devices see the inbound ringer: 1. The Uniden 5.8Ghz cordless phone base station that answers 95% of the calls 2. Asterisk with a win modem line card that: a. runs a perl AGI script to parse caller-id name and number b. rings a sip extension or c. answers the call and plays funny messages and DTMF tones at the telemarketers. Just make sure that Asterisk only RINGS the sip extensions but never sends the call to play a message or voicemail or any other Asterisk feature that will issue an implicit Answer and take the call. Yeah, except in the OP he mentions that he wants or is at least using Asterisk VM so your solution does not meet his needs. @~$7 for the privacy adaptor does not seem like overkill to me, at least price wise. Easy solution so OP can Can I have it both ways -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 asterisk on a shared line
If you don't have an objection to 24/7 then that is by far the best way, just get some fxs ports and each POTS phone can have it's own extension if you want. Certainly the way to go if there is no reason stopping you. On Thu, Jul 23, 2009 at 9:20 PM, Bill Lovett b...@ilovett.com wrote: I get how everything is connected with your setup, but if you pick up the cordless phone to answer a call does the sip extension just keep ringing until it times out? I like the exclusion adapter idea because it sounds like it would let me keep my dialplan intact. But I do take John and Trevor's point about putting everything through asterisk and running it 24/7. It would make things a lot simpler. On Jul 23, 2009, at 8:58 PM, Tom Browning wrote: An exclusion adapter is overkill. My Asterisk line card is the $10 Win modem card that I got from ebay. When you call my copper line, two devices see the inbound ringer: 1. The Uniden 5.8Ghz cordless phone base station that answers 95% of the calls 2. Asterisk with a win modem line card that: a. runs a perl AGI script to parse caller-id name and number b. rings a sip extension or c. answers the call and plays funny messages and DTMF tones at the telemarketers. Just make sure that Asterisk only RINGS the sip extensions but never sends the call to play a message or voicemail or any other Asterisk feature that will issue an implicit Answer and take the call. ___ -- 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 asterisk on a shared line
The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is not running, I can answer an incoming call from that phone. If asterisk is running, I can answer the call from a sip extension. Can I have it both ways? Can Asterisk back off if the secondary phone answers the call? Currently, if a call comes in and I answer it from the secondary phone Asterisk will continue to ring the sip extension and eventually drop into voicemail. You don't say how your Asterisk is currently connected to the PSTN line. Running Asterisk 24/7 would probably be the simplest. If adding some hardware is an option, it can be done by using an ATA for your secondary phone. Use an ATA that has both an FXO and FXS port, and bridges the two when power is lost. (such as the SPA-3102) You can either do some fancy wiring to power off the ATA when the asterisk box is shut down, or simply use a power bar. (shut down the asterisk box properly before killing it's power) Normal operation: PSTN line rings, Asterisk sees this and calls the sip extensions, including the sip extension of your secondary phone (via the ATA).The ATA will also see the line ringing on it's FXO port, but is configured to do nothing. Asterisk-off operation: PSTN line rings, asterisk is off so it does nothing, ATA is also off and has automatically bridged PSTN to the secondary phone. This approach also ensures that the secondary phone can still be used in emergencies if the Asterisk box has crashed, or during a power outage (* provided that the secondary phone does not, itself, require power to operate). ___ -- 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 asterisk on a shared line
Just a little clarification for people refering to Asterisk as a PBX and not an Answering Machine: In fact, Asterisk is neither a PBX nor an Answering Machine. Asterisk is a Telephony Toolkit. You can choose to use it as a PBX or an Answering Machine or both or even in some case as a something different than a PBX or Answering Machine. You should know that already, so this is just a reminder :-) Sent from my iPod On Jul 23, 2009, at 8:34 PM, Trevor Hammonds tre...@concipient.net wrote: Bill Lovett wrote: Can Asterisk be configured to hang up if another phone picks up? I'm a bit lost as far as terminology goes, but here's my setup. At home, I have asterisk answering calls from the pstn and sending them through to a sip extension or voicemail. All that is working fine. The box running Asterisk isn't on 24/7 so I have a secondary phone connected to the line as well. If Asterisk is not running, I can answer an incoming call from that phone. If asterisk is running, I can answer the call from a sip extension. Can I have it both ways? Can Asterisk back off if the secondary phone answers the call? Currently, if a call comes in and I answer it from the secondary phone Asterisk will continue to ring the sip extension and eventually drop into voicemail. Asterisk is a PBX, not an answering machine, so I would advise against this. It would be best to have Asterisk handle the phone line exclusively, 24/7. However, with that said, it is possible to accomplish what you are asking. Placing a telephone privacy/exclusion adapter on the line cord into Asterisk will cut off the phone line whenever a parallel telephone on the same line is picked up. This means that the instant you pick up any other phone on the line, it would cut off the line to Asterisk. Radio Shack used to sell a couple varieties of these. One was a two- way adapter with one side for phone and the other answering machine. You do not need to plug anything into the phone side for the device to work. The second device was just an inline exclusion device. I was unable to find these at Radio Shack's website. However, I found something similar at the following URLs: (See SER2A, SER2D, and SER3P at Sandman.com) http://www.sandman.com/lineshar.html http://www.trianglecables.com/telanmacorph.html http://www.iec-usa.com/cgi-bin/iec/COM9928 http://www.iec-usa.com/cgi-bin/iec/COM0006 Good luck! Sincerely, Trevor Hammonds ___ -- 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 asterisk on a shared line
Pascal Bruno wrote: Just a little clarification for people refering to Asterisk as a PBX and not an Answering Machine: In fact, Asterisk is neither a PBX nor an Answering Machine. Asterisk is a Telephony Toolkit. You can choose to use it as a PBX or an Answering Machine or both or even in some case as a something different than a PBX or Answering Machine. You should know that already, so this is just a reminder :-) Pascal, I agree with you that Asterisk is a telephony applications toolkit, and not a simple answering machine. However, Asterisk IS a PBX. The term answering machine in the context of this thread implies a device that has only basic answering functionality. Since Asterisk is capable of so much more than this basic functionality, I encouraged the OP to use it full time, rather than as an adjunct device. First line at: http://www.asterisk.org/ Asterisk is the world's leading open source PBX, telephony engine, and telephony applications toolkit. Sincerely, Trevor Hammonds ___ -- 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] best option for Conference timing with native Dahdi support
I asked this question a while back before Dahdi and have been using the X100P cards, but my understand is they will not have native support under Dahdi. What is the best option for installs that are pure SIP, but want to do reliable conferencing? Thanks! David Shauger Vice President Sollos Technology Solutions 678-317-9444 - voice 404-886-7603 - cell 772-679-5830 - fax d...@sollos.com http://www.sollos.com/ This email has been certified by Thawte Email certification helps prevent identity theft Virus scanning provided by Clam Antivirus ___ -- 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] odd behaviour with AGI and dial agent
Yes I have monitored it on the CLI and everything appears to work correctly but something is going wrong internally. I tried it with a php agi and it does work properly, so I guess it could be something to do with the fastagi. Even though the script is simple (at the moment) I would prefer to be able to use fastagi as the information is on another computer. By the way I am using asterisk 1.6.1 On 07/23/2009 11:00 PM, Danny Nicholas wrote: Have you monitored the call from CLI with verbose set up? What happens if you use regular AGI instead of FastAGI? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Keiron Liddle Sent: Wednesday, July 22, 2009 10:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] odd behaviour with AGI and dial agent Hi, I have come across an odd problem. Basically I am transferring a call to an agent. The agent is logged in and set as paused. In order to find which agent to call I am using a fastagi script to just set a variable. When it falls through the agi script and dials the agent (using the variable) it doesn't connect the call properly to the agent. I get the beep but no audio (along with some other strange behaviour with the channel not hanging up properly, core show channels doesn't work properly). Now if I just set the variable in the dialplan (ie. no agi), or just hardcode the agent being called then it works fine. It seems that calling the fastagi is doing something to the channel which means that it doesn't work properly afterwards. I have also tried calling the agent in the agi with the same problems. Does anyone have any idea what the agi script could be doing to the channel/call, what it could be changing and how I can make it work properly. Keiron ___ -- 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 asterisk on a shared line
That's right, they say it is a PBX because it is mostly used as such, but it is more than just a PBX. Some people use it as a VoiceMail tool or to handle just conference, some use it to add functionalities to other legacy PBX systems. Calling cards applications for example, a plain PBX wont be able to do that. Thats why I dont usually refer to it as a PBX. Pascal, I agree with you that Asterisk is a telephony applications toolkit, and not a simple answering machine. However, Asterisk IS a PBX. The term answering machine in the context of this thread implies a device that has only basic answering functionality. Since Asterisk is capable of so much more than this basic functionality, I encouraged the OP to use it full time, rather than as an adjunct device. First line at: http://www.asterisk.org/ Asterisk is the world's leading open source PBX, telephony engine, and telephony applications toolkit. Sincerely, Trevor Hammonds ___ -- 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 asterisk on a shared line
Yeah, except in the OP he mentions that he wants or is at least using Asterisk VM so your solution does not meet his needs. Ah, yes. My config would not allow Asterisk to be a part time voicemail destination. In my config, the POTS line has its own voicemail (it is actually a Comcast line and Comcast provided voicemail). I need Comcast provided voicemail to be the final destination on that line if noone answers and if the house is totally offline (power or broadband). (Comcast has bigger batteries than I do) ___ -- 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 asterisk on a shared line
I get how everything is connected with your setup, but if you pick up the cordless phone to answer a call does the sip extension just keep ringing until it times out? Actually no, the SIP extension stops ringing and Asterisk takes no further action. I like the exclusion adapter idea because it sounds like it would let me keep my dialplan intact. But I do take John and Trevor's point about putting everything through asterisk and running it 24/7. It would make things a lot simpler. A 24/7 box is great as long as you have some runtime on batteries to smooth out the occasional power failures and have a separate box to tinker with. If you are the only person expecting calls on that line, then a predictable result is less critical. (ie: Asterisk/voicemail is off and you don't answer and no other voicemail - ring no answer) ___ -- 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
