[asterisk-users] DECT Solution
Hello, I need to setup system of aroud 60 DECT phones with asterisk. So far I found http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710 However is there some cheap base station(similar to GSM cell) so I can handle all DECT phones centralized and plug it inside asterisk ? Thanks, Peter -- _ -- 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] How configure asterisk server extension.conf.
Hi, I have to create scenario like following, I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to Sip2sip.info , Can i able to communicate with this scenario ? How i should configure extention.conf in local asterisk sever to communicate with soft-phone which registered at Sip2sip.info ??Or if you have any other idea to crate such scenario please let me know ??Please also recommend me any good SIP Developer group ?? Best Regards,Sakharam Thorat. -- _ -- 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] Capture queue agent drop and put caller back in queue
2013/1/21 Mitch Claborn mitch...@claborn.net Asterisk 11 Occasionally we will have a partial power outage, or a piece of network equipment will fail, and our queue agents who are on active calls with callers will be disconnected from the caller. What I'd like to do is capture those calls and put them back in the queue (at a high priority) so that we don't lose the caller. I've tried to duplicate the situation in my lab: I have one agent in the queue, a caller dials into the queue, gets connected to the agent then I pull the ethernet cable out of the agent's computer (testing with a softphone) but I don't see anything happen on the asterisk console. core show channels shows the 2 channels still bridged even though the agent is gone. Shouldn't asterisk somehow know when the agent disappears? How can I accomplish my goal? I am not sure that from the PoV of the caller this solution would work - they would experience tens of seconds of silence plus they would have to go back to the queue. If this happens rarely, you could have a process call them back instead - you acknowledge what happened and have someone on-line with the person apologizing. We have a few clients implementing something like this for calls exiting the queue on timeouts and it seems to be well-liked by the callers. Of course it depends on what you are doing and the level of service that callers come to expect. Just my two cents, l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Integration with Social Media, Email and Web call center
And how would you have this working together with Asterisk queueing? I have seen solutions like this using agent pauses and then making everyithing happen outside the normal ACD flow, but it's a bit of a hack l. 2013/1/22 Danny Nicholas da...@debsinc.com For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Asterisk 11 / Missing Application SetCallerPres
Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI == Using SIP RTP CoS mark 5 -- Executing [100@sip:1] AGI(SIP/userid-001e, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPres) Options: (prohib_not_screened) [Jan 24 15:20:04] WARNING[15507][C-0030]: res_agi.c:2527 handle_exec: Could not find application (SetCallerPres) Why is the application not found, please? I think it should exist: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SetCallerPres Best regards, -Thorsten- -- _ -- 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] Asterisk 11 / Missing Application SetCallerPres
Simplest question first. Does it show up in core show applications or core show application SetCallerPres? If not, do a make menuselect and see if something broke in the ability to make the application. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Thursday, January 24, 2013 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 11 / Missing Application SetCallerPres Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI == Using SIP RTP CoS mark 5 -- Executing [100@sip:1] AGI(SIP/userid-001e, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPres) Options: (prohib_not_screened) [Jan 24 15:20:04] WARNING[15507][C-0030]: res_agi.c:2527 handle_exec: Could not find application (SetCallerPres) Why is the application not found, please? I think it should exist: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SetCaller Pres Best regards, -Thorsten- -- _ -- 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] Asterisk 11 / Missing Application SetCallerPres
Thanks! It is not activated. Also I found a comment there: Support Level: deprecated, Replaced by: func_callerid So I use this instead. Am 24.01.2013 15:33, schrieb Danny Nicholas: Simplest question first. Does it show up in core show applications or core show application SetCallerPres? If not, do a make menuselect and see if something broke in the ability to make the application. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner Sent: Thursday, January 24, 2013 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 11 / Missing Application SetCallerPres Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI == Using SIP RTP CoS mark 5 -- Executing [100@sip:1] AGI(SIP/userid-001e, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script Executing Application: (SetCallerPres) Options: (prohib_not_screened) [Jan 24 15:20:04] WARNING[15507][C-0030]: res_agi.c:2527 handle_exec: Could not find application (SetCallerPres) Why is the application not found, please? I think it should exist: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_SetCaller Pres Best regards, -Thorsten- -- _ -- 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] Is there a need to secure RTP ports?
Am 23.01.2013 um 18:33 schrieb Carlos Alvarez: On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk wrote: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has some trouble with my NAT). My question is - what are the vulnerabilities in this scenario at my end? I suppose some man-in-the-middle or eavesdropping attack is always a possibility - but that aside, is there anything that will attack RTP ports on Asterisk when there are no SIP ports open? I was looking into installing fail2ban - until I realised that there is no SIP port exposed for an attacker to poke at. I've been working in IP telephony for about ten years. I've never once heard of any attack on the RTP ports. While you can never say anything is impossible there's simply nothing listening on those ports. It's probably possible to have a DOS attack where someone starts sending RTP to all of your ports and they would interfere with a call, but they couldn't do more than that. That could work if your router has full cone NAT and a lot of other things fall into place. Still kind of out there as a real threat. -- Carlos Alvarez TelEvolve 602-889-3003 2 years ago someone demonstrated on the 27C3 in Berlin some interstings things you can do with RTP: http://media.ccc.de/browse/congress/2010/27c3-4193-en-having_fun_with_rtp.html (use the original file) Michael http://www.mksolutions.info smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] How to assign the button on the IP Phone to a feature?
Dear; Using Cisco IP Phones: How I can assign a button for a function. For example, if we pressed on this button, then we need to pickup the call from the group. Another thing: If the button pressed, then the call forward function to be enabled (and it should appear on the phone that it is grayed which indicates that it is enabled). So, when pressing this button, it means we are making the function ON and it is appearing at the LCD to be gray. And if we did another press on the button, then the gray to be removed and that means the function is OFF (so no call forward will happen). HOW? Generally speaking: How I can play in cisco button to assign them for features and controlling their appearance? Regards Bilal -- _ -- 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] Integration with Social Media, Email and Web call center
Yes it might be “hacky”, but anything that isn’t somewhat is going to come at a premium price. Today’s motto is “get her done as quick and cheap as possible”. It is a luxury to have a well-trained, professional staff providing solid solutions when folks want Top Quality at slave wage labor prices. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri Sent: Thursday, January 24, 2013 8:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Integration with Social Media, Email and Web call center And how would you have this working together with Asterisk queueing? I have seen solutions like this using agent pauses and then making everyithing happen outside the normal ACD flow, but it's a bit of a hack l. 2013/1/22 Danny Nicholas da...@debsinc.com For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] How to assign the button on the IP Phone to a feature?
On Thu, Jan 24, 2013 at 8:03 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dear; Using Cisco IP Phones: How I can assign a button for a function. For example, if we pressed on this button, then we need to pickup the call from the group. Which model line? The SPA series, or the 7900 and similar? They are completely different. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Integration with Social Media, Email and Web call center
They advised me to check jabber.org. Yes, jabber.org has a client that can send/receive and integrate with other social media (facebook, msn, twitter, ... etc). But, as an Agent who can login/logout and take a calls, how can I make it to be single login for voice and messages. So, if the agent is not available, he will not get a calls and will not get a messages. Those who used jabber.org or who used other than jabber.org for such requirement, what do you suggest? Regards Bilal -- For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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] Integration with Social Media, Email and Web call center
This is how I would see the process working 1. use curl/wget to query Facebook (etc.) 2. determine whether we are to drop a call into the queue or just process a message 3. determine agent availability through AMI process or asterisk -rx process. 4. drop the call into the queue or place the message if the agent is available 5. if the agent is unavailable, do alternate process. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, January 24, 2013 9:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Integration with Social Media, Email and Web call center They advised me to check jabber.org. Yes, jabber.org has a client that can send/receive and integrate with other social media (facebook, msn, twitter, ... etc). But, as an Agent who can login/logout and take a calls, how can I make it to be single login for voice and messages. So, if the agent is not available, he will not get a calls and will not get a messages. Those who used jabber.org or who used other than jabber.org for such requirement, what do you suggest? Regards Bilal -- For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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
[asterisk-users] g723 transcoding
It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that? -- _ -- 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] g723 transcoding
Hi Look at it this link http://asterisk.hosting.lv/ Kind Regards On Thu, Jan 24, 2013 at 10:34 AM, Richard Kenner ken...@gnat.com wrote: It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that? -- _ -- 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] Integration with Social Media, Email and Web call center
Hi, We before, used facebook graph api (json) on a php script. php would check new posts every minute, and write a new .call file into asterisk, with a sort of TTS call would go on queue, and once a member picks it up, he hears 'new facebook call from, bla bla, stating bla bla bla' He would then proceed to reply the facebook post (in our case also done in our software that would post back to FB via graph api) On 24 January 2013 15:28, Danny Nicholas da...@debsinc.com wrote: This is how I would see the process working 1. use curl/wget to query Facebook (etc.) 2. determine whether we are to drop a call into the queue or just process a message 3. determine agent availability through AMI process or asterisk -rx process. 4. drop the call into the queue or place the message if the agent is available 5. if the agent is unavailable, do alternate process. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, January 24, 2013 9:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Integration with Social Media, Email and Web call center They advised me to check jabber.org. Yes, jabber.org has a client that can send/receive and integrate with other social media (facebook, msn, twitter, ... etc). But, as an Agent who can login/logout and take a calls, how can I make it to be single login for voice and messages. So, if the agent is not available, he will not get a calls and will not get a messages. Those who used jabber.org or who used other than jabber.org for such requirement, what do you suggest? Regards Bilal -- For just the messaging part, you should be able to use wget or curl to interface and create messages. You might have to go a little higher level like C or Perl, but it sounds very doable. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, January 22, 2013 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integration with Social Media, Email and Web call center Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? Regards Bilal -- _ -- 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
[asterisk-users] question on SIP trunk and AMI to place call
When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? Thanks, jerry -- _ -- 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] question on SIP trunk and AMI to place call
Have you tried and looked up all events generated when you place the call? some of them are bound to have the variable callerid set On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote: When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? Thanks, jerry -- __**__**_ -- 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-**usershttp://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] g723 transcoding
On Thu, Jan 24, 2013 at 10:40:28AM -0500, Carlos Rojas wrote: Hi Look at it this link http://asterisk.hosting.lv/ Kind Regards On Thu, Jan 24, 2013 at 10:34 AM, Richard Kenner ken...@gnat.com wrote: It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that? If you do not want to arrange to pay your own royalties, a transcoding card like the TCE400B [1] can also handle transcoding to and from G.723.1. [1] http://www1.digium.com/en/products/telephony-cards/voice-compression/pci-express -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] question on SIP trunk and AMI to place call
Not the greatest solution, but since you are most likely using a script for the AMI process, you could do an Asterisk –rx “core show channels verbose”|grep SIP/testmachine-000d And get the dialed number from that. Actually you could issue the AMI command core show channels verbose. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Thursday, January 24, 2013 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call Have you tried and looked up all events generated when you place the call? some of them are bound to have the variable callerid set On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote: When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? Thanks, jerry -- _ -- 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] question on SIP trunk and AMI to place call
Have you tried and looked up all events generated when you place the call? some of them are bound to have the variable callerid set yes I have looked at all of them, CallerID is not set to the number I am calling. Jerry -- _ -- 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] question on SIP trunk and AMI to place call
Not the greatest solution, but since you are most likely using a script for the AMI process, you could do an Asterisk --rx core show channels verbose|grep SIP/testmachine-000d And get the dialed number from that. Actually you could issue the AMI command core show channels verbose. there is no core show channels verbose on Asterisk 11. There is on asterisk 1.4, core show channels on asterisk 11 has been changed. jerry -- _ -- 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] How to assign the button on the IP Phone
Both: SPA and 7900. let us say 7942. How? Regards Bilal Dear; Using Cisco IP Phones: How I can assign a button for a function. For example, if we pressed on this button, then we need to pickup the call from the group. Which model line? The SPA series, or the 7900 and similar? They are completely different. -- _ -- 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] How to assign the button on the IP Phone
On Thu, Jan 24, 2013 at 12:11 PM, bilal ghayyad bilmar...@yahoo.com wrote: Both: SPA and 7900. let us say 7942. How? Googled cisco 7942 soft keys, first result: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesoftk.html This is pretty off-topic, by the way. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] question on SIP trunk and AMI to place call
On 01/24/2013 10:46 AM, Jerry Geis wrote: When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. For example: exten = 500,1,Dial(SIP/digium02) Results in: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/10.x.x.x-0002 Destination: SIP/digium02-0003 CallerIDNum: 657-5309 CallerIDName: digium01 ConnectedLineNum: unknown ConnectedLineName: unknown UniqueID: Asterisk-01-1359052866.2 DestUniqueID: Asterisk-01-1359052866.3 Dialstring: digium02 -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] question on SIP trunk and AMI to place call
You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. Thanks, Jerry -- _ -- 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] question on SIP trunk and AMI to place call
This might have changed but IIRC /etc/asterisk/manager.conf controls what events you have access to. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, January 24, 2013 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. Thanks, Jerry -- _ -- 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] DECT Solution
On 01/24/2013 10:37 AM, Zyumbilev, Peter wrote: Hello, I need to setup system of aroud 60 DECT phones with asterisk. So far I found http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710 However is there some cheap base station(similar to GSM cell) so I can handle all DECT phones centralized and plug it inside asterisk ? Aastra has DECT base stations that can hook up to an Asterisk server. Last time I set one up it worked fine. You may want to try out several different brands of DECT phones and see which one the users like best. You don't want to get 50 support calls a day from your users complaining about how much the DECT phones suck. http://www.aastra.com/product-families.htm?curr_cat=DECT+Infrastructurecurr_type=Familymode_f=1mode_c=1mode_l=4 Polycom also has DECT stuff. I doubt it will come cheap. http://spectralink.polycom.com/dect_communications/index.html Regards, Patrick -- _ -- 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] DECT Solution
On Thu, Jan 24, 2013 at 12:52 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Polycom also has DECT stuff. I doubt it will come cheap. http://spectralink.polycom.**com/dect_communications/index.**htmlhttp://spectralink.polycom.com/dect_communications/index.html Not cheap, but this is the solution for large installations (over say 10 or 20 handsets or big spaces). Reliable, easy to work with. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] DECT Solution
I second what Carlos said. The Spectralink is the quality and reliability you want. On Jan 24, 2013 2:06 PM, Carlos Alvarez car...@televolve.com wrote: On Thu, Jan 24, 2013 at 12:52 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Polycom also has DECT stuff. I doubt it will come cheap. http://spectralink.polycom.**com/dect_communications/index.**htmlhttp://spectralink.polycom.com/dect_communications/index.html Not cheap, but this is the solution for large installations (over say 10 or 20 handsets or big spaces). Reliable, easy to work with. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
[asterisk-users] clicking sound with alaw codec
I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. The outgoing SDP looks like this: v=0 o=root 1691755711 1691755711 IN IP4 205.232.38.178 s=Asterisk PBX 10.7.1 c=IN IP4 205.232.38.178 t=0 0 m=audio 11432 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv The reply SDP is: v=0 o=default 1359060187 1359060187 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 Any suggestions on how to debug what's causing this? -- _ -- 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] clicking sound with alaw codec
Your sounds might be too loud. We use a lot of custom sounds here and when the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and clicks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner Sent: Thursday, January 24, 2013 2:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] clicking sound with alaw codec I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. The outgoing SDP looks like this: v=0 o=root 1691755711 1691755711 IN IP4 205.232.38.178 s=Asterisk PBX 10.7.1 c=IN IP4 205.232.38.178 t=0 0 m=audio 11432 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv The reply SDP is: v=0 o=default 1359060187 1359060187 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 Any suggestions on how to debug what's causing this? -- _ -- 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] clicking sound with alaw codec
Your sounds might be too loud. We use a lot of custom sounds here and when the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and clicks. Sorry I wasn't clear. This is *always*. I hear it over the call when there's talking and when there's dead silence (e.g., an empty MeetMe room). -- _ -- 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] Realtime vs Static Files
I am curious, is your version of asterisk correctly compiling the regserver field? Each server needs to have a distinct server name. Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as Registered on Asterisk1. -- _ -- 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] Realtime vs Static Files
2013/1/24 Dan Journo d...@keshercommunications.com I am curious, is your version of asterisk correctly compiling the regserver field? Each server needs to have a distinct server name. ** ** Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as Registered on Asterisk1. It is something really amazing... Can you run sip show peers on each one of the servers and post the response? You said the second asterisk is completely opaque to your peers. Can you run a tcpdump on secondary server to see if for some obscure reason the phones try to contact the secondary asterisk? Leandro -- _ -- 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] clicking sound with alaw codec
On 01/24/2013 09:44 PM, Richard Kenner wrote: [snip] When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come to mind in no particular order are: - DAHDI settings (sync source) - Asterisk server not properly grounded - timing is off (check logs) - shared interrupts (make sure nic/TDM card have their own) - jitterbuffer settings (try on/off) - echo cancellation going bonkers (OSLEC?) - QoS (proper priority for voice packets?) - PCI slot (if you have a card, try changing the slot it's in) Use Wireshark to see the difference between a good call and a bad one. If you see a lot of time jumps on the bad call then look at your network/QoS. Regards, Patrick -- _ -- 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] clicking sound with alaw codec
When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come to mind in no particular order are: Remember: this is only *one* particular SIP trunk. Use Wireshark to see the difference between a good call and a bad one. If you see a lot of time jumps on the bad call then look at your network/QoS. jumps? Note that a good call is G.729 and bad is G.711, so I wouldn't expect them to be at all similar. We throw a lot more bandwidth than even G.711 down the pipe between the two sites in terms of data each evening, so I don't think it's that kind of issue. I'm thinking in terms of distortion caused by transcoding someplace. -- _ -- 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] Integration with Social Media, Email and Web call center
On 22/01/2013, at 5:27 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; Can someone advise me where to find a technology (open source) that let us able to integrate with social media like whatsapp and facebook? And use this in call center (queuing the messages and routing it for agent)? Anyone give me a light to start? In the past I've sent calls to an agent in the queue with music on hold that contained a beep every 20 seconds (to remind them they're on a call) and then used the same code I do for screen popping to send them alternative records. I.E. web page, email, fax etc. It's stored in the database that that's what they were working on and then when they finished working on it they just hang up or press * to disconnect the call. That way you can use the standard Asterisk queues and they don't get bothered by anything else while they're working on it. Facebook might be a little harder as you wouldn't necessarily know when an incoming request came. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Integration with Social Media, Email and Web call center
On 24/01/2013, at 10:24 AM, bilal ghayyad bilmar...@yahoo.com wrote: They advised me to check jabber.org. Yes, jabber.org has a client that can send/receive and integrate with other social media (facebook, msn, twitter, ... etc). But, as an Agent who can login/logout and take a calls, how can I make it to be single login for voice and messages. So, if the agent is not available, he will not get a calls and will not get a messages. Those who used jabber.org or who used other than jabber.org for such requirement, what do you suggest? Ah, so yeah same as I said but when you get a jabber message you just send a call to the queue with one leg pointing to a particular music file and screen pop the relevant data. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] DAHDI: How to supress notification of changing CallerID on transfer?
Hi Richard, the macro you linked to did the trick for me - thank you! Greetings from Wuppertal Max Grobecker Am 24.01.2013 00:18, schrieb Richard Mudgett: - Original Message - Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 - IVR answers and puts caller to the chosen queue - Someone picks up the phone (Internal ext. 321) - CallerID shown on customers device changes to 020212345-321 Same when I park the call and pick it up on another phone. I don't want this to happen and can't figure out how to disable this on DAHDI or at least the current channel. I tried facilityenable=no in chan_dahdi.conf, but this only supresses signalling the on hold status. We are using a german ISDN Anlagenanschluss (bri_cpe) whith DDI served by the Deutsche Telekom, connected to a ISDN card which is used with DAHDI. Is there a hidden config flag or something to disable this for DAHDI? Or maybe a channel variable to temporarily disabling this on some channels? chan_dahdi in Asterisk 11 has this option to easily do what you want: ; Send ISDN conected line information. ; ; block: Do not send any connected line information. ; connect: Send connected line information on initial connect. ; update: Same as connect but also send any updates during a call. ; Updates happen if the call is transferred. (Default) ; ;colp_send=update You can also use the interception macros on v1.8. https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Richard -- _ -- 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 -- -- - Portunity GmbH - Werner-Seelenbinder-Str. 23 -- 42477 Radevormwald - Germany - - Portal: http://www.portunity.de - - General: Phone: +49 (0)202 - 69555 - 0 - eMail/SIP: i...@portunity.de - Fax: +49 (0)202 - 69555 - 190 - - Support: Phone: +49 (0)202 - 69555 - 300 - eMail/SIP: supp...@portunity.de - - Amtsgericht Koeln HRB 38162 - USt-Identnummer DE206277867 - Geschaeftsfuehrung: Bjoern Ruecker, Bernd Schnell -- -- _ -- 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] clicking sound with alaw codec
- jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371424, src=RTP [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371434, src=RTP -- _ -- 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] question on SIP trunk and AMI to place call
On 01/24/2013 01:13 PM, Jerry Geis wrote: You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. The Dial events are created by app_dial. So long as you are using app_dial to create your outbound channel, you should have that event. Channel technology shouldn't matter. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] clicking sound with alaw codec
On 01/24/2013 11:57 PM, Richard Kenner wrote: - jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371424, src=RTP [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371434, src=RTP Check https://issues.asterisk.org/jira/browse/ASTERISK-12042 Regards, Patrick -- _ -- 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] clicking sound with alaw codec
Check https://issues.asterisk.org/jira/browse/ASTERISK-12042 I did. But that was with an unofficial G.729. This is with the supplied alaw codec. -- _ -- 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] Mail list settings?
- Original Message - From: Bryant Zimmerman brya...@zktech.com Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. It's operating as it always has. Mailman is set to insert the list address in the Reply-To field without stripping the original. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.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
