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Hello, I have been asked if it is possible for recorded calls to be flagged with a decriptive title automatically and filed. The application is a third party call center operation, where certain customers that engage the call center want all calls recorded and then a submission of call samples at the end of the calling period. This is being done by hand at the moment. I fully expect that I

...t;) in new stack -- Executing [8100 at test:2] Set("SIP/1001-00000000", "CALLERID(all)=""") in new stack -- Executing [8100 at test:3] PrivacyManager("SIP/1001-00000000", "") in new stack -- <SIP/1001-00000000> Playing 'privacy-unident.g729' (language 'en') -- <SIP/1001-00000000> Playing 'privacy-prompt.g729' (language 'en') voip*CLI> Disconnected from Asterisk server /usr/sbin/safe_asterisk: line 146: 21648 Segmentation fault -- Thanks, Phil

this purpose. I have been reading the voip-info pages and have set up AJAM and can get results from doing http requests to the asterisk server, however, this is in the form of an action, such as login, rather than subscribing to an event. I have been looking here for information http://www.voip-info.org/wiki/view/asterisk+manager+events Does anyone know if subscriptions is possible with AJAM?

I'm sure this is a topic that's come up plenty of times before, but is it possible to implement native parallelization in rsync? ?There are several blog posts listing workarounds, but they are largely unideal. My initial thought would be that rsync could split up the file list across N workers; the incremental file list would still be generated as it does today, but can then be divided

Hi, I''m using the new nested forms functionality and I having problems with new created elements. I have an invoice with lines. I would like to be able to create a new invoice, in the same form insert new lines dinamically and saving the invoice and the lines when everything is validated, not before. The problem is that when I add a line dinamically to the form, the new line created

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

...in new stack -- Executing NoOp("Zap/7-1", "") in new stack -- Executing PrivacyManager("Zap/7-1", "") in new stack == Parsing '/etc/asterisk/privacy.conf': == Parsing '/etc/asterisk/privacy.conf': Found -- Playing 'privacy-unident' (language 'en') -- Playing 'privacy-prompt' (language 'en') -- Playing 'privacy-thankyou' (language 'en') -- Changed Caller*ID to "Privacy Manager" <6145551212> -- Executing LookupBlacklist("Zap/7-1", "&quot...

...CallerID returned with error on channel 'Zap/1-1' -- Executing Zapateller("Zap/1-1", "") in new stack -- Executing PrivacyManager("Zap/1-1", "") in new stack == Parsing '/etc/asterisk/privacy.conf': Found -- Playing 'privacy-unident' -- Playing 'privacy-prompt' Is there a way to NOT have asterisk go through my s,2 if CID is an error? Or maybe change my incoming so nothing happens if CID returns with an error? extensions.conf [incoming] exten => s/,1,Zapateller exten => s,1,NoOp exten => s,2,Priv...

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

Hello Everyone, I have an installation where the client has a Microsoft SQL database that holds all of their case information. They would like the asterisk system to require users to enter a valid case number when making an outgoing call. I?m seeing some documentation regarding people using Microsoft SQL for CDR storage, however nothing regarding validating authentication using a Microsoft SQL

I'm still trying to get an idea about how best to handle patches on CentOS. Say I want to apply security patches automatically on a nightly basis. But when the push from 4.2 to 4.3 comes around, I want to defer that for when I can do it manually. Is that possible? (Preferably with yum, but I would use up2date if that were necessary.) Also, I'm used to doing 'yum update'. I

I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

...ayback(welcome) >exten => 077777,4,GotoIf($[foo${CALLERIDNUM} = foo]?7:5) >exten => >077777,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?calle rid=${CALLERIDNUM}) >exten => 077777,6,Hangup >exten => 077777,7,Wait(2) >exten => 077777,8,Playback(privacy-unident) >exten => 077777,9,Hangup > >exten => 2500,1,Dial(Zap/32,40) >exten => 2500,2,VoiceMail2(u2500) >exten => 2500,3,Hangup >exten => 2500,102,VoiceMail2(b2500) >exten => 2500,103,Hangup > >exten => 2501,1,Dial(Zap/33,40) >exten => 2501,2,VoiceMail...

Hello, This afternoon I received the new that my institution, in the main direction, is planning (for political reasons, I don't know which one??) migrate to Debian. I have a hotted conversation with my boss cause I've been working on CentOS during near a year and half. I explained it that both distros just implement the protocols that are standards, that could be installed a web server

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

I've a call center using IAX softphones provided by a third party. We've observed problems where the IAX phones seem unable to use our PRI trunks. A sample anonymized call is provided below with the PRI debug calls embedded. Any thoughts, comments or suggestions would be welcome. In anonymizing it, I preseved the format and number of digits sent. -- Accepting AUTHENTICATED