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After Hangup call contrtol in dialplan

January 8, 2014, 12:42 am
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Is there any way to control call after one side Hangup() in dialplan............

Actually this is a call center scenario ::

customer call comes in through pstn , our agent attends the call and talks and at the end agent cuts the call, the customer (caller) is still in the call and he is then routed to another IVR or some thing. I just tried with the Hangup handlers....not getting the result.
here is my code:
Code: pstn call is routed to this Hangup-test context-
[hangup-test]

exten => 999,1,NoOp(for hangup test)
         same => n,Saynumber(5500)
         same => n,Dial(SIP/2000,20) ;;;;;agent attends and hangsup the call
         same => n,Set(CHANNEL(hangup_handler_push)=hdlr3,s,1(args));

[hdlr3]

exten => s,1,Verbose(0, Executed Third)
         same => n,Playback(tt-monkeys)
         same => n,Dial(SIP/4000,20)




let me know if there is an alternate way for this also.....I'm a newbie in this
Thanks in advance Image

Statistics : Posted by vinc • on Wed Jan 08, 2014 2:42 am • Replies 1 • Views 43
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Asterisk featuremap +Recording issue

January 7, 2014, 10:46 pm
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Asterisk featuremap +Recording issue

January 7, 2014, 10:46 pm
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I believe I have found the problem, it seams like Mixmonitor doesn't continue recording the message because it sees the blind transfer from ## as 2 divert call because both calls have different uniqueid.

Caller A phone Caller B then caller B blind transfers to Caller C: (## transfer)
Caller A to Caller B -
Uniqueid: 1389173420.258
CDR(uniqueid): 1389173420.258
Recording folder: /var/spool/asterisk/monitor/1389173420.258.WAV (Both recordings are linked)
Caller B to Caller C -
Uniqueid: 1389173420.258
CDR(uniqueid): 1389173420.260
Recording folder: /var/spool/asterisk/monitor/1389173420.260.WAV (Only Second call)

Caller A phone Caller B then caller B blind transfers to Caller C: (button transfer)
Caller A to Caller B -
Uniqueid: 1389173770.287
CDR(uniqueid): 1389173770.287
Recording folder: /var/spool/asterisk/monitor/1389173770.287.WAV (Both calls together)
Caller B to Caller C -
Uniqueid: 1389173770.287
CDR(uniqueid): 1389173770.287
Recording folder: Same
_---------------------------------------_
Problem call:
Caller A phone Caller B then caller A blind transfers to Caller C: (## transfer)
Caller A to Caller B -
Uniqueid: 1389173483.263
CDR(uniqueid): 1389173483.263
Recording folder: /var/spool/asterisk/monitor/1389173483.263.WAV (Only first call)
Caller A to Caller C -
Uniqueid: 1389173488.266
CDR(uniqueid): 1389173483.263
Recording folder: None

As we can see in the colour match, if the uniqueid match on both calls MixMonitor continues to keep the recording stream open. However when the uniqueid doesn't match MixMonitor doesn't keep the stream open. However these 2 calls now have the sanem CDR(uniqueid) which makes my macro no record a file.

However should this be logged as a Bug? I believe asterisk to be creating the ## blind transfer as 2 different calls rather then having then bridged or linked.

Normal blind transfers from Buttons on the phone works correctly, will it not be possible to have asterisk process the ## transfers the same?

Statistics : Posted by faqterson • on Wed Jan 08, 2014 12:46 am • Replies 3 • Views 74
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Asterisk featuremap +Recording issue

January 7, 2014, 10:46 pm
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Caller A phone Caller B then caller B blind transfers to Caller C: (## transfer)
Caller A to Caller B -
Uniqueid: 1389173420.258
CDR(uniqueid): 1389173420.258
Recording folder: /var/spool/asterisk/monitor/1389173420.258.WAV (Both recordings are linked)
Caller B to Caller C -
Uniqueid: 1389173420.258
CDR(uniqueid): 1389173420.260
Recording folder: /var/spool/asterisk/monitor/1389173420.260.WAV (Only Second call)

Caller A phone Caller B then caller B blind transfers to Caller C: (button transfer)
Caller A to Caller B -
Uniqueid: 1389173770.287
CDR(uniqueid): 1389173770.287
Recording folder: /var/spool/asterisk/monitor/1389173770.287.WAV (Both calls together)
Caller B to Caller C -
Uniqueid: 1389173770.287
CDR(uniqueid): 1389173770.287
Recording folder: Same

Caller A phone Caller B then caller A blind transfers to Caller C: (button transfer)
Caller A to Caller B -
Uniqueid: 1389173839.292
CDR(uniqueid): 1389173839.292
Recording folder: /var/spool/asterisk/monitor/1389173839.292.WAV (First part)
Caller A to Caller C -
Uniqueid: 1389173847.295
CDR(uniqueid): 1389173839.294
Recording folder: /var/spool/asterisk/monitor/1389173839.294.WAV (Second part)

_---------------------------------------_
Problem call:
Caller A phone Caller B then caller A blind transfers to Caller C: (## transfer)
Caller A to Caller B -
Uniqueid: 1389173483.263
CDR(uniqueid): 1389173483.263
Recording folder: /var/spool/asterisk/monitor/1389173483.263.WAV (Only first call)
Caller A to Caller C -
Uniqueid: 1389173488.266
CDR(uniqueid): 1389173483.263
Recording folder: None

No second part to the call.

Regards

Statistics : Posted by faqterson • on Wed Jan 08, 2014 12:46 am • Replies 3 • Views 74
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Asterisk featuremap +Recording issue

January 7, 2014, 10:46 pm
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Please can I ask for same assistance:
[featuremap]
blindxfer => ##
atxfer => **

1. Incoming call transferred with ## records the hole recording in one file, doesn't stop.
2. Internal call from one extension to another continues the call recording in one file doesn't stop.
3. Outgoing call transferred stops recording on first transfer. (Tried the Append but the original recording gets created as soon as the transfer takes place)

My recording macro:
[macro-check-record]
exten => s,1,GotoIf($[${RECORD}=YES]?2:5)
exten => s,2,Set(EXISTS=${STAT(e,/var/spool/asterisk/monitor/${CDR(uniqueid)}.wav)})
exten => s,3,GotoIf(${EXISTS}?5)
exten => s,4,MixMonitor(${CDR(uniqueid)}.wav,b,/usr/local/bitco/convert ${CDR(uniqueid)})
exten => s,5,NoOp(RECORD: ${RECORD})

What I have found with Option 3:
An outgoing call when transferred doesn't create a new uniqueid for the transferred call. Which cause my Macro to skip MixMonitor.

Caller A phone Caller B, Caller A then transfer the call to Caller C; The call between Caller B and Caller C stops Recording

Regards

Statistics : Posted by faqterson • on Wed Jan 08, 2014 12:46 am • Replies 3 • Views 74
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Digium G100 Reviews?

January 8, 2014, 9:08 am
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ehufford,

I am not a Digium Employee Image

The G100's and G200's are nice devices.

First of the all the GUI is interactive and one of the best I have seen and immediately provides confidence in the product. We configured and brought the first unit up in less than 30 mins with no manuals.

As for operation, we have several now out there with clients and all do their job nicely and as stated. The routing configuration is easy, and the ability to capture packets and pri logs makes it generally easy to work issues out.

I like the Patton devices as well, but as you said, a lot to configure, but the Patton allows a lot more tweaking, so it has its uses as well, but If I was onsite, putting a Gateway in, on the fly, I would rather be doing it with a Digium Gateway.

The only thing that has bugged me was the support from Digium compared to the support from Patton. But to be fair, Digium finally comes through with a bit of poking. Only being honest in the review.

Regards

Bob

Statistics : Posted by ehufford • on Wed Jan 08, 2014 11:08 am • Replies 2 • Views 44
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Digium G100 Reviews?

January 8, 2014, 9:08 am
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Digium G100 Reviews?

January 8, 2014, 9:08 am
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Has anyone used the the Digium G100 gateway units?

I'm looking for a standalone T1/PRI to SIP appliance, and I'd like to use a G100 because I like Asterisk (and have an Asterisk server running our system), but I can't find many reviews on them. I've had people recommend Patton to me, but I'm a little reluctant because the configuration is always a pain.

Any experiences with the Digium hardware, good or bad?

Statistics : Posted by ehufford • on Wed Jan 08, 2014 11:08 am • Replies 2 • Views 44
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Call drops only on a call transfer

February 14, 2013, 4:28 pm
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Have you solved this issue.

I have a client experiencing the issue.

Calls get dropped only if they are transferred to him. usually 30 seconds after the transfer.


I know this post is pretty old.. but do remember what you did about this.

Statistics : Posted by mamunoz • on Thu Feb 14, 2013 6:28 pm • Replies 5 • Views 1038
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Call drops only on a call transfer

February 14, 2013, 4:28 pm
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David55 thanks again for continuing to help me, I really appreciate it.

I am confused about your last statement in the reply, did you not see the link to the traces that I linked or are you saying they aren't correct, I am confused.

Mark

Statistics : Posted by mamunoz • on Thu Feb 14, 2013 6:28 pm • Replies 5 • Views 1038
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Call drops only on a call transfer

February 14, 2013, 4:28 pm
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You are not doing a transfer; you are setting up a tandem connection. The only way that would change the RTP handling is if you had directmedia=yes, but, if you had that, Asterisk would not be expecting to see any RTP, so wouldn't be timing it out.

Even if your inline trace is only partial, and is actually the enquiry leg of an attended transfer, Asterisk still routes the call as though it had been a simple through connection. (I suppose in that case, you might just have had a system that worked because you were using directmedia, but are no longer using it. Direct media would also remove the effects of a VM on the through audio.))

If I have to work to get at a trace, I'll normally not bother.

Statistics : Posted by mamunoz • on Thu Feb 14, 2013 6:28 pm • Replies 5 • Views 1038
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Call drops only on a call transfer

February 14, 2013, 4:28 pm
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I do have some traces I ran from my troubleshooting the ring group issue.

Here is a link to the 3 traces I ran while testing.

http://wikisend.com/download/376122/CallCapture.zip

We have been using asterisk on this EC2 instance now for well over 6 months without any issues of call quality etc. I am not sure if that is what you're referencing or not by your mentioning of it however. Either way it works flawless except for this transfer issue. Everything I have researched says that RTP activity is a result of a firewall issue it just doesn't make sense as a call that isn't transferred works as expected. I would expect if it were a firewall issue this wouldn't be the case, but that's why I am posting on these forums though haha because I am at a loss. Image

Statistics : Posted by mamunoz • on Thu Feb 14, 2013 6:28 pm • Replies 5 • Views 1038
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Call drops only on a call transfer

February 14, 2013, 4:28 pm
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Asterisk has no concept of IVRs or ring groups. These are constructs created by dialplans. You need to ask the person who wrote the dialplan, e.g. http://www.freepbx.org if you are using the FreePBX GUI.

Asterisk transfers completely remove the call from Asterisk, so it is unlikely that you are really doing a tarnsfer.

It looks like you have a NAT or firewall problem that is physically blocking the the RTP, but there is far too little information to say much more than that. You need to provide a trace showing the SDP exchange.

I don't know enough about the EC2 environment, except that it is a virtual machine system, and therefore not designed for the tight real time constraints of VoIP, so likely to produce high jitter and choppy voice announcements.

Statistics : Posted by mamunoz • on Thu Feb 14, 2013 6:28 pm • Replies 5 • Views 1038
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Call drops only on a call transfer

February 14, 2013, 4:28 pm
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So my setup is running on Amazon EC2 instance. Currently I have ALL ports being forwarded so essentially no firewall configured.

I can make and receive calls perfectly fine.

I noticed in testing this setup I had an issue that if a user called the IVR then the IVR rang a ring group it would ring all involved but the person that answered would hear nothing and neither would the other end. The calls didn't disconnect just no audio. If I tried to call again every member of the ring group would ring except the people that have last tried to answer a call from that ring group. Almost as if Asterisk still thought they were on the call and didn't ring the extension because of that. Long story short we don't use ring groups so I never worried about it.

Today a user called AT&T, they were walking through their IVR and it went to transfer them to a support agent and as soon as AT&T initiated the transfer you hear to faint clicks and thats it. The call timer still runs so Asterisk didn't end the call but I could no longer hear anything. Looking in the logs you see this.

Code: -- Called SIP/BW-SIP-A/+18003310500
    -- SIP/BW-SIP-A-00000317 is making progress passing it to SIP/120-00000316
    -- SIP/BW-SIP-A-00000317 is making progress passing it to SIP/120-00000316
    -- SIP/BW-SIP-A-00000317 answered SIP/120-00000316
[2013-02-14 17:01:25] NOTICE[1934]: chan_sip.c:25735 check_rtp_timeout: Disconnecting call 'SIP/BW-SIP-A-00000317' for lack of RTP activity in 31 seconds
[2013-02-14 17:01:26] NOTICE[1934]: chan_sip.c:25735 check_rtp_timeout: Disconnecting call 'SIP/120-00000316' for lack of RTP activity in 31 seconds
[2013-02-14 17:01:31] WARNING[1934]: chan_sip.c:3873 __sip_autodestruct: Autodestruct on dialog '4279fcaf-30ae2d0c-54cf9b69@10.0.2.5' with owner in place (Method: BYE)
[2013-02-14 17:01:58] WARNING[1934]: chan_sip.c:3873 __sip_autodestruct: Autodestruct on dialog '11c873e076a0e6627612f4fc09f84ded@107.22.237.xxx:5060' with owner in place (Method: BYE)


Which turns out to be the EXACT same errors I was getting months ago when testing the ring groups.

I use FreePBX 2.10 as a GUI and Bandwidth.com as my SIP provider.

SIP Settings has NAT set to yes, it has external IP proper and local networks as 10.0.0.0/255.0.0.0

Reinvite is set to Update and RTP is set to 30 seconds

I compiled Asterisk from source here is version output

Asterisk 1.8.12.0 built by root @ domU-12-31-39-06-2E-05 on a x86_64 running Linux on 2012-05-22 23:25:36 UTC

I am running Amazon Linux 64 bit 2012.03

Help with this would be greatly appreciated and please let me know if more info is needed.

Statistics : Posted by mamunoz • on Thu Feb 14, 2013 6:28 pm • Replies 5 • Views 1038
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Codec Issue (?)

January 9, 2014, 9:41 am
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You can't connect a text message to a voice channel. You have to use the message sending applications instead.

Statistics : Posted by zachakit86 • on Thu Jan 09, 2014 11:41 am • Replies 1 • Views 30
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Codec Issue (?)

January 9, 2014, 9:41 am
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Okay,

I'm having a really odd issue....I am currently using XMPP to receive unsolicited messages, no problem there, I send a message to my asterisk box via xmpp, and it comes it to the dialplan and jumps to a context as I want, however when I try to initial the Dial(SIP/1234@context) it doesn't work...I get this error:

Executing [s@macro-smsdial:2] Dial("Message/ast_msg_queue", "SIP/context/1234") in new stack
[Jan 9 11:29:32] NOTICE[17857][C-00000000]: chan_sip.c:29701 sip_request_call: Asked to get a channel of unsupported format (nothing) while capability is (gsm|ulaw|alaw)
[Jan 9 11:29:32] WARNING[17857][C-00000000]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available)

What gives? I've tried SoftHangUp(Message), it will not go to an h extension of I use Hangup(), as I am thinking the Channel that is "open" is strictly for SMS/XMPP/SIP Messages....

Any ideas would be great!

Statistics : Posted by zachakit86 • on Thu Jan 09, 2014 11:41 am • Replies 1 • Views 30
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Uk ISDN and Early Media settings ?

January 9, 2014, 6:34 am
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Hi there, I am currently running Asterisk 1.8.24.0 with Dahdi drivers DAHDI Version: 2.7.0.1.


I am trying to get early media working, Sip to Sip is fine but when a call comes in via ISDN no Audio is heard, even though asterisk show the media is being played.

Here is my settings
chan_dahdi.conf
Code: [channels]
language=en
usecallerid=yes
cidsignalling=v23
cidstart=polarity
nationalprefix=44
internationalprefix=+
callprogress=yes
progzone=uk


sip.conf
Code: [general]
directmedia=no
dtmfmode=rfc2833
rtptimeout=600
jbenable=yes
prematuremedia=yes
progressinband=yes


extension.conf
Code: exten => _1510,1,progress()
exten => _151,2,Wait(1)
exten => _151,3,Playback(tt-monkeys,noanswer)
exten => _151,4,Hangup()nf


pri dedug
Code:
PRI Span: 8
PRI Span: 8 < Protocol Discriminator: Q.931 (8)  len=40
PRI Span: 8 < TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent from originator)
PRI Span: 8 < Message Type: SETUP (5)
PRI Span: 8 < [a1]
PRI Span: 8 < Sending Complete (len= 1)
PRI Span: 8 < [04 03 90 90 a3]
PRI Span: 8 < Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0  Info transfer capability: 3.1kHz audio (16)
PRI Span: 8 <                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
PRI Span: 8 <                                User information layer 1: A-Law (35)
PRI Span: 8 < [18 03 a9 83 81]
PRI Span: 8 < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0  Exclusive  Dchan: 0
PRI Span: 8 <                       ChanSel: As indicated in following octets
PRI Span: 8 <                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
PRI Span: 8 <                       Ext: 1  Channel: 1 Type: CPE]
PRI Span: 8 < [1e 02 85 81]
PRI Span: 8 < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the remote user (5)
PRI Span: 8 <                               Ext: 1  Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ]
PRI Span: 8 < [6c 0c 21 83 37 39 31 37 33 30 31 34 32 36]
PRI Span: 8 < Calling Party Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
PRI Span: 8 <                                 Presentation: Presentation allowed, Network provided (3)  '7917301426' ]
PRI Span: 8 < [70 04 80 31 35 31]
PRI Span: 8 < Called Party Number (len= 6) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '151' ]
PRI Span: 8 -- Making new call for cref 1
PRI Span: 8 Received message for call 0x2b8ed81194b0 on link 0x2b8eb0035600 TEI/SAPI 0/0
PRI Span: 8 -- Processing Q.931 Call Setup
PRI Span: 8 -- Processing IE 161 (cs0, Sending Complete)
PRI Span: 8 -- Processing IE 4 (cs0, Bearer Capability)
PRI Span: 8 -- Processing IE 24 (cs0, Channel ID)
PRI Span: 8 -- Processing IE 30 (cs0, Progress Indicator)
PRI Span: 8 -- Processing IE 108 (cs0, Calling Party Number)
PRI Span: 8 -- Processing IE 112 (cs0, Called Party Number)
PRI Span: 8 q931.c:8646 post_handle_q931_message: Call 1 enters state 6 (Call Present).  Hold state: Idle
Span 8: Processing event PRI_EVENT_RING
PRI Span: 8 q931.c:5704 q931_call_proceeding: Call 1 enters state 9 (Incoming Call Proceeding).  Hold state: Idle
PRI Span: 8
PRI Span: 8 > DL-DATA request
PRI Span: 8 > Protocol Discriminator: Q.931 (8)  len=10
PRI Span: 8 > TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent to originator)
PRI Span: 8 > Message Type: CALL PROCEEDING (2)
PRI Span: 8 TEI=0 Transmitting N(S)=97, window is open V(A)=97 K=7
PRI Span: 8
PRI Span: 8 > Protocol Discriminator: Q.931 (8)  len=10
PRI Span: 8 > TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent to originator)
PRI Span: 8 > Message Type: CALL PROCEEDING (2)
PRI Span: 8 > [18 03 a9 83 81]
PRI Span: 8 > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0  Exclusive  Dchan: 0
PRI Span: 8 >                       ChanSel: As indicated in following octets
PRI Span: 8 >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
PRI Span: 8 >                       Ext: 1  Channel: 1 Type: CPE]
    -- Accepting call from 'Mymobile' to '151' on channel 0/1, span 8
    -- Executing [151@PBX-IN:1] Progress("DAHDI/i8/Mymobile-af3", "") in new stack
PRI Span: 8
PRI Span: 8 > DL-DATA request
PRI Span: 8 > Protocol Discriminator: Q.931 (8)  len=9
PRI Span: 8 > TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent to originator)
PRI Span: 8 > Message Type: PROGRESS (3)
PRI Span: 8 TEI=0 Transmitting N(S)=98, window is open V(A)=97 K=7
PRI Span: 8
PRI Span: 8 > Protocol Discriminator: Q.931 (8)  len=9
PRI Span: 8 > TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent to originator)
PRI Span: 8 > Message Type: PROGRESS (3)
PRI Span: 8 > [1e 02 81 88]
PRI Span: 8 > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
PRI Span: 8 >                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
    -- Executing [151@PBX-IN:2] Wait("DAHDI/i8/Mymobile-af3", "1") in new stack
    -- Executing [151@PBX-IN:3] Playback("DAHDI/i8/Mymobile-af3", "tt-monkeys,noanswer") in new stack
    -- <DAHDI/i8/Mymobile-af3> Playing 'tt-monkeys.gsm' (language 'en')
PRI Span: 8
PRI Span: 8 < Protocol Discriminator: Q.931 (8)  len=13
PRI Span: 8 < TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent from originator)
PRI Span: 8 < Message Type: DISCONNECT (69)
PRI Span: 8 < [08 02 80 90]
PRI Span: 8 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: User (0)
PRI Span: 8 <                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
PRI Span: 8 < [1e 02 81 88]
PRI Span: 8 < Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  Location: Private network serving the local user (1)
PRI Span: 8 <                               Ext: 1  Progress Description: Inband information or appropriate pattern now available. (8) ]
PRI Span: 8 Received message for call 0x2b8ed81194b0 on link 0x2b8eb0035600 TEI/SAPI 0/0
PRI Span: 8 -- Processing IE 8 (cs0, Cause)
PRI Span: 8 -- Processing IE 30 (cs0, Progress Indicator)
PRI Span: 8 -- Found active call: 0x2b8ed81194b0 cref:1
PRI Span: 8 q931.c:9084 post_handle_q931_message: Call 1 enters state 12 (Disconnect Indication).  Hold state: Idle
Span 8: Processing event PRI_EVENT_HANGUP_REQ
    -- Span 8: Channel 0/1 got hangup request, cause 16
  == Spawn extension (PBX-IN, 151, 3) exited non-zero on 'DAHDI/i8/Mymobile-af3'
PRI Span: 8 q931.c:7135 q931_hangup: Hangup other cref:1
PRI Span: 8 q931.c:6892 __q931_hangup: ourstate Disconnect Indication, peerstate Disconnect Request, hold-state Idle
PRI Span: 8 q931.c:5930 q931_release: Call 1 enters state 19 (Release Request).  Hold state: Idle
PRI Span: 8
PRI Span: 8 > DL-DATA request
PRI Span: 8 > Protocol Discriminator: Q.931 (8)  len=9
PRI Span: 8 > TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent to originator)
PRI Span: 8 > Message Type: RELEASE (77)
PRI Span: 8 TEI=0 Transmitting N(S)=99, window is open V(A)=99 K=7
PRI Span: 8
PRI Span: 8 > Protocol Discriminator: Q.931 (8)  len=9
PRI Span: 8 > TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent to originator)
PRI Span: 8 > Message Type: RELEASE (77)
PRI Span: 8 > [08 02 81 90]
PRI Span: 8 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Private network serving the local user (1)
PRI Span: 8 >                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
    -- Hungup 'DAHDI/i8/Mymobile-af3'
PRI Span: 8
PRI Span: 8 < Protocol Discriminator: Q.931 (8)  len=5
PRI Span: 8 < TEI=0 Call Ref: len= 2 (reference 1/0x1) (Sent from originator)
PRI Span: 8 < Message Type: RELEASE COMPLETE (90)
PRI Span: 8 Received message for call 0x2b8ed81194b0 on link 0x2b8eb0035600 TEI/SAPI 0/0
PRI Span: 8 q931.c:8944 post_handle_q931_message: Call 1 enters state 0 (Null).  Hold state: Idle
PRI Span: 8 q931.c:7135 q931_hangup: Hangup other cref:1
PRI Span: 8 q931.c:6892 __q931_hangup: ourstate Null, peerstate Null, hold-state Idle
PRI Span: 8 Destroying call 0x2b8ed81194b0, ourstate Null, peerstate Null, hold-state Idle
Span 8: Processing event PRI_EVENT_HANGUP_ACK


I have placed calls from mobiles and landlines. I get the same result as the debug, media is played but it cannot be heard.

Statistics : Posted by RZer0 • on Thu Jan 09, 2014 8:34 am • Replies 1 • Views 29
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Uk ISDN and Early Media settings ?

January 9, 2014, 6:34 am
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I very much doubt that UK ISDN would support early media if you are not a licensed network operator; it is open to abuse for free of charge information calls.

Statistics : Posted by RZer0 • on Thu Jan 09, 2014 8:34 am • Replies 1 • Views 29
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Can Asterisk route 911 to landline when using Google Voice?

January 10, 2014, 4:54 am
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If you want to dial a 911 from a separate provider or line, then you just need to make a separate rule in the outgoing part of your dialplan. Very simple.

If you want to send the 911 calls via a crapy provider because your are to cheap to afford yourself an emergency/backup line or a VoIP provider account - sorry, can't help you there. I rather stick to solutions that work.

Statistics : Posted by ccherng • on Fri Jan 10, 2014 6:54 am • Replies 2 • Views 54
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Can Asterisk route 911 to landline when using Google Voice?

January 10, 2014, 4:54 am
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Google Voice does not properly support 911 calls. Is there a way to configure Asterisk to handle 911 calls properly.

I read that the Obi 110 device can do this with a setup where it is connected to your internet router and also connected to the phone landline. Normal calls are sent by the Obi 110 to Google Voice via your router as usual. But 911 calls are directed by the Obi 110 to the landline. This should make it possible to avoid having to pay for a landline at all since 911 calls do not require a paid landline. Is that correct?

Is there a way to do this kind of setup with a Linux server running Asterisk. The Linux server would have two ethernet ports. One connected to the router as usual and the second connected to the phone landline using some kind of adapter (is it ATA?).

Statistics : Posted by ccherng • on Fri Jan 10, 2014 6:54 am • Replies 2 • Views 54
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