Re: [asterisk-users] How to tell if there is a transfer from CDR?
Hi, I use CEL or Call Event Logging in 1.8 to get a more concise picture of what happened in a call. We use it for a bunch of stuff including billing attended and unattended transfers differently. If you are thinking of upgrading, it's worth a try. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: 05 September 2010 03:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the 2nd record would include the time of the 1st record as well. So if part one took 20 seconds and part 2 40 seconds, then the 2nd record would have 60 seconds as billable. The only workaround was to check the BLINDTRANSFER var and reset cdr if it was populated. Please members of this list, I would love to hear more input as I'm sure this has changed. Also I would not be surprised that I'm wrong in my analysis as more than 4 years has passed since and I might have forgotten. TIA On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote: Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] How to tell if there is a transfer from CDR?
On blind transfers I believe the two cdr's have the same unique id . On
attended transfers there is no real way I have found to address this issue.
CDR's with transfers really suck the way they are right now. On blind transfers
you can do some flagging of the second CDR by checking in your dialing contexts
to confirm it is a blind transfer ${BLINDTRANSFER}. On attended transfers you
are just out of luck. You have to sort them out with CDR's. This cost us some
money with inbound toll free calls because we did not know this occurred this
way for some time.
Bryant
From: C F shma...@gmail.com
Sent: Saturday, September 04, 2010 10:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR?
Last time I analyzed this (I believe back in 1.2) there was no way of
telling. However a blind transfered call would generate 2 CDR
recoreds:
1. For the part of the call with the transferrer and transfered.
2. For the part of the call with the transferee and transfered.
The call duration for the 2nd record would include the time of the 1st
record as well. So if part one took 20 seconds and part 2 40 seconds,
then the 2nd record would have 60 seconds as billable.
The only workaround was to check the BLINDTRANSFER var and reset cdr
if it was populated.
Please members of this list, I would love to hear more input as I'm
sure this has changed. Also I would not be surprised that I'm wrong in
my analysis as more than 4 years has passed since and I might have
forgotten.
TIA
On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote:
Is there any way to know if a call was transferred from reading the
CDR? Any relation in fields like UNIQUEID? Something that can be
scripted to make a special report?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] How to tell if there is a transfer from CDR?
Nic How stable is 1.8 really? It sounds like you are running it in production is this the case? CDR Transfer issues and rfc2833 DTMF issues are hitting us hard with 1.6.2.x. We want to move as soon as 1.8 is stable enough. Thanks Bryant From: Nic Colledge n...@njcolledge.net Hi, I use CEL or Call Event Logging in 1.8 to get a more concise picture of what happened in a call. We use it for a bunch of stuff including billing attended and unattended transfers differently. If you are thinking of upgrading, it's worth a try. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: 05 September 2010 03:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the 2nd record would include the time of the 1st record as well. So if part one took 20 seconds and part 2 40 seconds, then the 2nd record would have 60 seconds as billable. The only workaround was to check the BLINDTRANSFER var and reset cdr if it was populated. Please members of this list, I would love to hear more input as I'm sure this has changed. Also I would not be surprised that I'm wrong in my analysis as more than 4 years has passed since and I might have forgotten. TIA On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote: Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- _ -- 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] Registering and initiating a SIP call without a SIP client
Can I generate SIP registration and call from Asterisk without a SIP client? I need to initiate a call from asterisk and play a recorded message. Gautam -- _ -- 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] Registering and initiating a SIP call without a SIP client
Yes. Send your code. Consider using call files. Here is a part of what works for me. [-system] exten = s,1,Answer exten = s,n,Wait(2) exten = s,n,Playback(pa-welcome) please record your broadcast after the beep ;exten = s,n,Playback(beep) exten = s,n,Wait(1) exten = s,n,Record(/var/lib/asterisk/sounds/en/record713.gsm) ;exten = s,n,Record(LINDA_RISTIG_linda005) ; record this: this welcome to dial a restaurant ??? ;exten = s,n,Wait(1) exten = s,n,Background(pa-confirm) ; press 1 to send or zero to hangup exten = s,n,WaitExten(10) ;exten = s,n,Hangup() exten = 1,1,System(cp /etc/asterisk/pizza/*.call /tmp/) exten = 1,n,System(mv /tmp/*.call /var/spool/asterisk/outgoing/) exten = 0,1, Hangup() ;; ;; [pizza] exten = 13,1,Answer() exten = 13,n,Wait(1) exten = 13,n,Playback(record713) ;exten = 13,n,Playback(LINDA_RISTIG_IVR) ;exten = 13,n,Playback(calleveryone) ;exten = 13,n,WaitExten(5) exten = 13,n,Goto(13,1) On Sun, Sep 5, 2010 at 6:20 PM, Gautam Desai gdesai...@yahoo.com wrote: Can I generate SIP registration and call from Asterisk without a SIP client? I need to initiate a call from asterisk and play a recorded message. Gautam -- _ -- 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] Registering and initiating a SIP call without a SIP client
Am 06.09.2010 00:20, schrieb Gautam Desai: Can I generate SIP registration and call from Asterisk without a SIP client? I need to initiate a call from asterisk and play a recorded message. Gautam hello, have a look at the sip.conf.sample file how to register asterisk as a sip client and also at callfiles or the cli command originate. best regards steve -- _ -- 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] Registering and initiating a SIP call without a SIP client
It can be done either using a call file and a clever dial plan or via the manager interface, again with a clever dialplan On 09/05/2010 03:20 PM, Gautam Desai wrote: Can I generate SIP registration and call from Asterisk without a SIP client? I need to initiate a call from asterisk and play a recorded message. Gautam -- _ -- 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] evil disconnect of call with cisco 1760
On 9/4/2010 1:31 AM, Jeremy Kister thought: On 8/29/2010 3:25 AM, Jeremy Kister wrote: whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a critical packet being missed. hmm, either no one has any clue/suggestions or they just don't care about the issue - I better figure it out myself. I wonder if it has to do do with progress indicators on the 1760. Thanks Jeremy, that was it! I ended up putting: progress_ind progress enable 8 progress_ind connect enable 8 on each dial-peer pots, and then: progress_ind setup enable 3 progress_ind connect enable 8 on each dial-peer voip. Problem seems solved. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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 tell if there is a transfer from CDR?
Bryant, We have been using a pre-1.8 trunk version of asterisk that has been pretty stable for us. We have a fairly small user base currently and decided to take the risk with a trunk version after some testing basically because of the availability of CEL as it lets us do a bunch of things we couldn't do with 1.6 CDR. I have briefly tested the beta3 of 1.8 and it seemed ok but were holding off for the release version (with a if it ain't broke don't fix it mentality). In our environment and my limited experience 1.8 is shaping up to be a great release (Great work guys, thanks to everyone working on asterisk!) I recommend you fire it up somewhere to test and see if you still have the issues, CEL can be pretty verbose and confusing at first but it does give you a lot more information about what happened during a call and when. On the other hand you may see it working and decide it's not for you. As for DTMF we only have a few IVR Menu style interfaces that don't currently see much use so I can't really give a definitive answer, but we have not had any problems with it. Worst case scenario, you test it and find some problems. I can't speak for the developers, but while it's in beta, now's the time to find them! Regards, Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: 05 September 2010 21:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Nic How stable is 1.8 really? It sounds like you are running it in production is this the case? CDR Transfer issues and rfc2833 DTMF issues are hitting us hard with 1.6.2.x. We want to move as soon as 1.8 is stable enough. Thanks Bryant From: Nic Colledge n...@njcolledge.net Hi, I use CEL or Call Event Logging in 1.8 to get a more concise picture of what happened in a call. We use it for a bunch of stuff including billing attended and unattended transfers differently. If you are thinking of upgrading, it's worth a try. Nic. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: 05 September 2010 03:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to tell if there is a transfer from CDR? Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the 2nd record would include the time of the 1st record as well. So if part one took 20 seconds and part 2 40 seconds, then the 2nd record would have 60 seconds as billable. The only workaround was to check the BLINDTRANSFER var and reset cdr if it was populated. Please members of this list, I would love to hear more input as I'm sure this has changed. Also I would not be surprised that I'm wrong in my analysis as more than 4 years has passed since and I might have forgotten. TIA On Fri, Sep 3, 2010 at 5:06 PM, Carlos Chavez cur...@telecomabmex.com wrote: Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- _ -- 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
