[asterisk-users] [Zaptel] numberplan-local context from nowhere?
Hello Using Asterisk 1.4.20 and Zaptel 1.4.3-1, I notice that Asterisk sends FXO calls to a context named numberplan-local that is not mentionned in my configuration file, which prevents incoming calls to be successfull: === /etc/asterisk/zapata.conf == [trunkgroups] [channels] ;Send incoming calls to this context in extensions.conf context=from_fxo switchtype=national usecallerid=no hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes canpark=yes busydetect=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no signalling = fxs_ks channel = 1 === /etc/asterisk/extensions.conf == [general] writeprotect = yes autofallthrough=yes ... [from_fxo] ;Context set in zapata.conf exten = s,1,Verbose(In fxo) === Console: zap show channels == Chan Extension Context Language MOH Interpret pseudonumberplan-locadefault 1numberplan-locadefault === Console: incoming call on FXO == -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at numberplan-local,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at numberplan-local,s,1 still failed so falling back to context 'default' [Jan 1 01:06:08] WARNING[628]: pbx.c:2483 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' = Has someone seen this before and knows why Asterisk comes up with that numberplan-local context? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Zaptel] numberplan-local context from nowhere?
On Sat, 12 Feb 2011 10:14:42 +0100, Gilles codecompl...@free.fr wrote: Using Asterisk 1.4.20 and Zaptel 1.4.3-1, I notice that Asterisk sends FXO calls to a context named numberplan-local that is not mentionned in my configuration file, which prevents incoming calls to be successfull: Found what it was: This context was used in the samples, and I missed user.conf. Renaming this file and rebooting the host solved the problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables losing their value????
Apologies, using two underscores (I retested) did not cause the error
On Sat, Feb 12, 2011 at 1:42 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Alrighty Gents, let's see if any of you have encountered this
one...Variables losing their value...I'm setting a variable with four
underscores (used to be two, had same issue) so it can be inherited by child
channels, and then the next line in the dialplan I use it but it appears to
be empty...I've googled and found nothing stating this kind of weirdness..
Asterisk 1.8.2.2 (upgrading to 1.8.2.3 shortly)
dialplan:
[menu.main]
exten = s,1,Set(recfile=${FILTER(0-9,${UNIQUEID})});
exten = s,n,Set(logfile=${recfile}) ;
The log output:
-- Executing [s...@menu.main:1] Set(SIP/-,
recfile=12974953060) in new stack
-- Executing [s...@menu.main:2] Set(SIP/-, logfile=)
in new stack
Anybody have thoughts?
Thanks,
S McGowan
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[asterisk-users] SIP Hardphone that works well with asterisk
Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
zyxel On Sat, Feb 12, 2011 at 4:01 PM, ast guy ast...@gmail.com wrote: Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
On Sat, Feb 12, 2011 at 9:31 AM, ast guy ast...@gmail.com wrote: Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag Currently most every phone works well, if the patch for Cisco subscriptions gets tested then I would say any non-skype phone. I have a snom 360, a snom m3 and a snom 870 on my desks and work with polycom phones when the market dictates. Cisco 79XX and newer phones work fine until you get to the more advanced features like directed pickup and SLA but that support is coming. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
Aastra Polycom because they can be configured using a TFTP server. Great for large installations with centralized management. Mitel 5215/5224 because they are dead simple to configure (via web gui) and just plain work with no screwing around. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ast guy Sent: Saturday, February 12, 2011 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Hardphone that works well with asterisk Hi, I have been out of touch with asterisk for quit some time and needed some recommendations. I am looking for SIP hardphone that works well with asterisk server. Pls suggest. cheers /ag -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
On Sat, Feb 12, 2011 at 9:50 AM, Terry Brummell te...@brummell.net wrote: Aastra Polycom because they can be configured using a TFTP server. Great for large installations with centralized management. Mitel 5215/5224 because they are dead simple to configure (via web gui) and just plain work with no screwing around. Terry Phone Provisioning is a part of Asterisk. It works over HTTP now and with an FTP or TFTP proxy can work over multiple protocols at once. Read More: https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk I added example snom support and will have to start a review board for adding Cisco, Aastra and others. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
Yes, I use provisioning for my Polycom's. And unfortunately, as far as I know, the Mitel's do not support tftp/http provisioning. I did just upgrade my 5215's to SIP Rel8 and I see them do a call to /init in the tftp, but I don't know what the phone is trying to do in that folder. Anyway, that's taking this off topic of the OP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Saturday, February 12, 2011 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Hardphone that works well with asterisk Terry Phone Provisioning is a part of Asterisk. It works over HTTP now and with an FTP or TFTP proxy can work over multiple protocols at once. Read More: https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk I added example snom support and will have to start a review board for adding Cisco, Aastra and others. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
Thanks for the comments, I will go through the detail and price and then will buy accordingly, cheers /ag On Sat, Feb 12, 2011 at 2:11 PM, Terry Brummell te...@brummell.net wrote: Yes, I use provisioning for my Polycom's. And unfortunately, as far as I know, the Mitel's do not support tftp/http provisioning. I did just upgrade my 5215's to SIP Rel8 and I see them do a call to /init in the tftp, but I don't know what the phone is trying to do in that folder. Anyway, that's taking this off topic of the OP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Saturday, February 12, 2011 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Hardphone that works well with asterisk Terry Phone Provisioning is a part of Asterisk. It works over HTTP now and with an FTP or TFTP proxy can work over multiple protocols at once. Read More: https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk I added example snom support and will have to start a review board for adding Cisco, Aastra and others. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
On Sat, Feb 12, 2011 at 10:11 AM, Terry Brummell te...@brummell.net wrote: Yes, I use provisioning for my Polycom's. And unfortunately, as far as I know, the Mitel's do not support tftp/http provisioning. I did just upgrade my 5215's to SIP Rel8 and I see them do a call to /init in the tftp, but I don't know what the phone is trying to do in that folder. Anyway, that's taking this off topic of the OP. Actually its not very far off topic. When starting a new project, say 1000+ extensions, the decision is complex. If you need 1000+ IP phones you need to verify your supply chain and calculate cost of installation. If you have manually touch or even scan the phones then things get complex real quick. For example some suppliers will print additional information on each box of phones to make deployment easier. If you get your boxes of phones with user or cubicle names printed in relation to the MAC and also have the list for configuration then you look like a super hero... Price is not the deciding point on real projects, its the supporting infrastructure. We are also working to get some ATAs like the Audiocodes to work from the res_phoneprov. I have hacked together a TFTP proxy with the help of some existing projects. You can read more here http://etel.wiki.oreilly.com/wiki/index.php/Dynamic_Phone_Provisioning_with_res_phoneprov_and_TFTP Some notes on IP hardphones to look for... * Button Labels, When dealing with languages other than English you need to verify that the phones are available with per language button labels or no labels and on screen soft buttons in the language you need. * Configuration of VLAN on the phone. Can it be done automatic or is it a manual setting. Does the phone support vlan tagging. * Firmware updates: Are they free, fast and stable? * POE standards. Does the phone communicate its required power to the switch correctly (not all switches or phones like each other.). Example: Polycom 330 and Cisco 2960 = all ports think they need 15.4 watts and may over saturate the switch power supply causing random ports to have POE turn off. Polycom 331 and Cisco 2960 = All ports communicate with the the phone and set wattage to ~4 watts +- cable length * Sidecars for the phones. Can you dynamically add side cars to the phones. Lucky that today you can just order snom Visions and do some hacking to fix this issue. * Directory XML App. When you exceed the internal directory size, can you create an XML app to allow the searching of a corporate directory from the phones screen. (Hint: res_phoneprov + xml app + dynamic alphabetic pages) * Power Adapter: Many providers have order numbers with and without power adapters. In every installation you will need power adapters for testing, verifying and for edge devices. * Headsets: Customers request computability with headsets. There are Linksys phones shipping today with bluetooth. My snom 870 has two types of headset plugs on the back. * Handset cord length: Some phones have long handset cords, others not so much. * Handset weight, does it feel too light and weak * Mounting, Are there wall mounting and multiple position desk mounting options * Wireless/dect/cordless = Is it a dangerous environment? I have purchased simple analog phones at walmart for projects where the cordless phones are damaged too often. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium te220
On 02/11/2011 07:56 AM, Albert wrote: can anymore drop me a asterisl's config for digium te220b (with ec) or at least some good tutorial of configuratin e1 line with that card ? The information you are looking for (for Asterisk, not for 'asterisl') is provided in the manual for the card; if you didn't receive a printed copy when you purchased it, you can read it online on www.digium.com. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using files .call or AMI
Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered.I did the page and everything but when I do the clicktodial I dont know how transfer the call to this agent. I ask the extension and user before login so I know what agent is in each extension to transfer the call to rigth agent. Anybody can give an idea ?TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using files .call or AMI
as you know you have 2 ways. using ami or .call files. if you have experience, the AMI is more powerful. you must have a context in your extensions.conf to manage agent procedures, it looks like a simple context, that you must have, for managing queues. with .call file or ami dial your customers, () and divert it to the defined context for queue. for example test.call Channel: SIP/customer number@your carrier Context: your queue context . ask if you need more info best On Sat, Feb 12, 2011 at 7:53 PM, Edwin Quijada listas_quij...@hotmail.comwrote: Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenario I have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered. I did the page and everything but when I do the clicktodial I dont know how transfer the call to this agent. I ask the extension and user before login so I know what agent is in each extension to transfer the call to rigth agent. Anybody can give an idea ? TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using files .call or AMI
On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote: I have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered. Usually it's the other way round: the agent's phone rings, and when he picks it up the other end gets dialled. That's trivial with call files: Channel: (local channel ID for agent) Context: (context for calling local channel) Extension: (remote party's phone number) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using files .call or AMI
Date: Sat, 12 Feb 2011 21:35:29 + From: ro...@firedrake.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote: I have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered. Usually it's the other way round: the agent's phone rings, and when he picks it up the other end gets dialled. That's trivial with call files: Channel: (local channel ID for agent) Context: (context for calling local channel) Extension: (remote party's phone number) This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? TIA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer Device Data
Hello!
I am trying to find out the device name and/or other identifying data
to be used in a context when a device transfers the call to new a
phone number. From running tests, it looks like the account code
variable (${CDR(accountcode)}) is set to the account code of the
device that placed the original call, so if the callee device (not the
original calling device) is making the transfer to a new number, the
account code will not be correct, since it will be the account code of
the calling device, but not the called device.
How do I find out which device is making the transfer?
Thanks for any suggestions!
Elliot
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Re: [asterisk-users] Using files .call or AMI
My problem is that I dont know how to do for transfer the call to agentExample, I have this .call Channel: Zap/g1/8652323454MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: call-file-test Extension: 10 So my context is this [call-file-test ]exten = 10,1,Dial(SIP/2031,tT)exten = 10,2,hangup In this case I call the number 8652323454 if the call is connect this call in the context call-file-test uisng extension 10 for tranfering this call to extension 2031, but this doesnt work. The call file works fine but when I try to transfer the call I get an error Any help ? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: l...@lopl.net Date: Sat, 12 Feb 2011 21:22:50 +0330 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI as you know you have 2 ways. using ami or .call files. if you have experience, the AMI is more powerful. you must have a context in your extensions.conf to manage agent procedures, it looks like a simple context, that you must have, for managing queues. with .call file or ami dial your customers, () and divert it to the defined context for queue. for example test.call Channel: SIP/customer number@your carrier Context: your queue context. ask if you need more infobest On Sat, Feb 12, 2011 at 7:53 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered. I did the page and everything but when I do the clicktodial I dont know how transfer the call to this agent. I ask the extension and user before login so I know what agent is in each extension to transfer the call to rigth agent. Anybody can give an idea ?TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using files .call or AMI
On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? I'm using this to place calls from local IP-phones over the PSTN. So my script will generate, say: Channel: SIP/lanphone Context: from-lan Extension: 08001234567 taking the 0800... from the list of customer details. SIP/lanphone is the ID of the originating phone. Extension is the sequence the agent would dial if he were placing the call himself. The originating phone rings; when it's picked up, the Asterisk server calls the Extension number and bridges the two calls, so the local agent hears ringing tones from the far end. All the agent has to do is pick up the phone when it rings and put it down when the call is over. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Device Data
${BLINDTRANSFER} should hold the device name of the one doing the
blind transfer.
On Sat, Feb 12, 2011 at 6:06 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello!
I am trying to find out the device name and/or other identifying data
to be used in a context when a device transfers the call to new a
phone number. From running tests, it looks like the account code
variable (${CDR(accountcode)}) is set to the account code of the
device that placed the original call, so if the callee device (not the
original calling device) is making the transfer to a new number, the
account code will not be correct, since it will be the account code of
the calling device, but not the called device.
How do I find out which device is making the transfer?
Thanks for any suggestions!
Elliot
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[asterisk-users] Fax for Asterisk SIP-TDM
Is it possible to do SIP-Asterisk-TDM in a single step with FFA? Or does FFA always use TIFF files? I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102 ATA's at the fax machines and send faxes directly over a PRI. Mark -- Mark Willis Star One Telecom Office: 1-800-889-7001 Cell: 210 880 5097 http://staronetel.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Files, Variable passing
Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, Action: login\r\n); fputs($oSocket, Events: off\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Exten: 3001\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Priority: $strPriority\r\n); fputs($oSocket, MaxRetries: $strMaxReTry\r\n); fputs($oSocket, RetryTime: $strRetryTime\r\n); fputs($oSocket, SetVar: DIAL1=$number1\r\n); fputs($oSocket, SetVar: DIAL2=$number2\r\n); fputs($oSocket, SetVar: AcceptParallel=$ap\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); Here I am trying to set three variables but they do not seem to be passed on to the extensions for dialing Am I following the right syntax ? Thanks -dani -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
