[asterisk-users] Regrading Speech Recognition.
Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads & Thanks Akhilesh -- _ -- 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] basic sip quesiton
On Jul 4, 2012, at 9:20 PM, Thomas Perron wrote: > What am I missing please? sip show registry shows that I am registered. What are you missing? A question, or at the least, a description of whatever problem you are having? Also, a meaningful subject that somewhat talks to the content of your question. -- _ -- 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] basic sip quesiton
What am I missing please? sip show registry shows that I am registered. [general] register => 5552530146:tam...@sip3.voipvoip.com ; ; [sip3.voipvoip.com] bindport=5060 ;you can use different port if the default is blocked bindaddr=0.0.0.0 ;binds to all ;this is for codec negotiation between the useragent and asterisk disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm context=incoming ;default context where incoming calls are passed. this should be the context where your local user.s extensions reside [outbound-trunk] ;this is the second section of you sip.conf file. Here you can create your trunk through which you will throw your outgoing calls to axvoice. host=sip3.voipvoip.com type=peer dtmfmode=rfc2833 canreinvite=yes -- _ -- 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] Speech recognition in asterisk using google voice API
On 07/04/2012 08:44 PM, Bruce B wrote: > Hey Zaf, > > Just checking the Google Speech Recognition package again and I can't see > WolframAlpha.agi file. I check all of your projects on Git hub but can't > find wolframalpha.agi. Please let us know what the URL is. > > Thanks, > Bruce > It is under the folder samples/wolfram/ https://github.com/zaf/asterisk-speech-recog/tree/master/samples/wolfram Lefteris Zafiris -- _ -- 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] Outbound Asterisk calls default directmedia specifications
Thanks for the response.. I did change it in the [general] settings.My setup is something like I have a remote conference (not meetme) which will send reinvite to redirect the RTP flow to a different server to load balance.There are three clients who join in the conference and i can listen to two other clients speak from the third client but when i record the conversation my recording of one of the clients ends before the stipulated hangup time. I am guessing this is because one of the clients doesn't understand what to do with a reinvite.. Any suggestions.In the SIP.conf i have changed the directmedia option to no and also enabled the ignoresdpversion option. On Tue, Jul 3, 2012 at 10:01 PM, SamyGo wrote: > I don't think you can set SIP properties in some variables anywhere in > asterisk dialplan or call file. What you can do is change the directmedia > options of the SIP or any other channel you're using. i.e if your call file > has > > CHANNEL=SIP/12345@latestgateway > > Then change the properties of the [latestgateway] in sip.conf. Also if > you're using an IP address directly > > CHANNEL=SIP/12345@10.10.4.4 > > Then you can change the directmedia directive in sip.conf [general] > settings. > > Hope it helped. > > BR > Sammy Go. > > On Wed, Jul 4, 2012 at 2:08 AM, sathiish kumar > wrote: > >> I am using call files to make calls to a remote machine but can't seem >> to quite understand the directmedia options that are set by default in >> Asterisk.Is there any way i can specify the directmedia options using call >> files? >> >> >> >> -- >> _ >> -- 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 > -- Sathiish Kumar Mohan Kumar Technical Operations Intern, Citrix Online 7408 Hollister Avenue,Goleta, Santa Barbara CA-93106 - -- _ -- 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 1.8.13 PlayTones App
Hi,
i'm trying to implement a Playtones App into my IVR. If a invalid numer
is entered, Asterisk should play the info tone defined at the
indications.conf.
My Extensions.conf
exten => i,1,Set(CHANNEL(language)=de)
exten => i,2,Progress()
exten => i,3,PlayTones(info)
exten => i,n,Playback(fettefinger)
exten => i,n,Wait(2)
exten => i,n,StopPlayTones()
exten => i,n,Goto(i,3)
The Console shows
Invalid extension '5' in context 'support' on SIP/200-000a
== CDR updated on SIP/200-000a
-- Executing [i@support:1] Set("SIP/200-000a",
"CHANNEL(language)=de") in new stack
-- Executing [i@support:2] PlayTones("SIP/200-000a", "info") in
new stack
-- Executing [i@support:3] Playback("SIP/200-000a",
"fettefinger") in new stack
-- Playing 'fettefinger.alaw' (language 'de')
-- Executing [i@support:4] Wait("SIP/200-000a", "2") in new stack
But the only thing i can hear is my Soundfile fettefinger.
BR Jakob
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Re: [asterisk-users] Speech recognition in asterisk using google voice API
Hey Zaf, Just checking the Google Speech Recognition package again and I can't see WolframAlpha.agi file. I check all of your projects on Git hub but can't find wolframalpha.agi. Please let us know what the URL is. Thanks, Bruce On Thu, Jan 12, 2012 at 2:49 PM, Lefteris Zafiris wrote: > On 01/12/2012 05:50 PM, Danny Nicholas wrote: > > Two more "offerings" - #1 - add DTMF parameter so function can be > stopped by > > pressing a digit or digits other than * or # - #2 - add an option to > > "silence" the beep. If you were using this in an IVR and wanted to say > > "press 1 or say help for help", silencing the beep before recording > would > > (IMO) make the rendering sound more "professional"/less "mechanical". > > Both features added: > > - > Usage > - > agi(speech-recog.agi,[lang],[timeout],[intkey],[NOBEEP]) > Records from the current channel untill the timeout (set to 10 seconds > by default, -1 for no timeout) is reached or the interrupt key (# by > default) is pressed. > If NOBEEP is set, no beep sound is played back to the user to indicate > the start of the recording. > > There is now also the option to enable SSL for encrypted communication > between your pbx and the google voice server. > > Updated code can be found here: > https://github.com/zaf/asterisk-speech-recog/tarball/master > > > Lefteris Zafiris > > -- > _ > -- 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] Timer1 RFC and SIP.CONF
Hello, I am trying to get clarity with the sip.conf timer configuration. The current configuration states: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 However, according to RFC 3261: (EXCERPT 17.1.1.1) T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms. Nearly all of the transaction timers described here scale with T1, and changing T1 adjusts their values. The request is not retransmitted over reliable transports. After receiving a 1xx response, any retransmissions cease altogether, and the client waits for further responses. The server transaction can send additional 1xx responses, which are not transmitted reliably by the server transaction. Eventually, the server transaction decides to send a final response. (EXCERPT 13.2.2.4 2xx Responses) The UAC core considers the INVITE transaction completed 64*T1 seconds after the reception of the first 2xx response. According to the RFC, the 64*t1 timeout is not for provisional responses, but for final responses. This seems to be in contradiction to what is stated in the sip.conf file. Thanks, Elliot -- _ -- 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] Queue Member login from IAX trunk
hmm before I think of any solution: All I can tell you is that if you're
trying to use ;
exten => 105,n,Set(AGENT_SIP=${*DB(IAX2/intranet/agent_ip)*})
Both your Asterisk's server will need to have this DB in common/accessible.
Got it.. Use realtime Queues on Server-A ;
when Agent Logs in to Server B put values of the new Agent in MySQL
queue-memebers table:
Only Server-A needs to be in realtime. Server-B will only be inserting and
deleting agents from the table.
I hope it works.
Sammy
On Wed, Jul 4, 2012 at 2:57 PM, Jakob-Matthias Böttger wrote:
> mhh that makes sense indeed. My intention is to add Agents wich are
> connected to serverb to the queue at servera. But i don't know how to tell
> the queue that the Agent ist reachable at IAX/serverb/agent_sip
>
> so the extension should be something like
>
> exten => 105,n,Read(AGENT_SIP,agent-newlocation)
> exten => 105,n,Set(AGENT_SIP=${DB(IAX2/intranet/agent_ip)})
>
> so the agent enters the number from the phone he is connected. Then
> Asterisk adds "IAX2/serverb" to the number and saves it as agend phone
> number...
>
> Regards Jakob
>
>
> Am 04.07.2012 11:45, schrieb SamyGo:
>
> Hi,
>
> exten => 105,n,Read(AGENT_SIP,agent-newlocation)
> exten => 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})
>
> Above two lines are very suspicious: AGENT_SIP is a variable which is
> getting some DTMF from caller. agent-newlocation is the message you
> want to be played while getting the AGENT_SIP input.
> What is the next line doing in SET() !!? Please explain that.
>
> Regards,
> Sammy
>
> On Wed, Jul 4, 2012 at 2:32 PM, Jakob-Matthias Böttger wrote:
>
>> Hello
>>
>> i got two Asterisk servers connected with IAX2.
>>
>> At server 1 i hosted a queue named support. Now i want the Agent to login
>> from SIP Phones Connected to server B.
>> Therefore i wrote a extension like
>>
>> exten => 105,1,Authenticate(1234)
>> exten => 105,n,AddQueueMember(support,,)
>> exten => 105,n,Read(AGENT_SIP,agent-newlocation)
>> exten => 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})
>> exten => 105,n,Playback(agent-loginok)
>> exten => 105,n,Playback(vm-goodbye)
>> exten => 105,n,Hangup
>>
>> Now server A takes IAX/serverB as argument for AGENT_SIP but deletes the
>> entered number in agent-newlocation. Do you have any ideas how to solve
>> that?
>>
>> Best regards Jakob
>>
>>
>> --
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>>http://www.asterisk.org/hello
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
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>
>
>
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Re: [asterisk-users] Queue Member login from IAX trunk
mhh that makes sense indeed. My intention is to add Agents wich are
connected to serverb to the queue at servera. But i don't know how to
tell the queue that the Agent ist reachable at IAX/serverb/agent_sip
so the extension should be something like
exten => 105,n,Read(AGENT_SIP,agent-newlocation)
exten => 105,n,Set(AGENT_SIP=${DB(IAX2/intranet/agent_ip)})
so the agent enters the number from the phone he is connected. Then
Asterisk adds "IAX2/serverb" to the number and saves it as agend phone
number...
Regards Jakob
Am 04.07.2012 11:45, schrieb SamyGo:
Hi,
exten => 105,n,Read(AGENT_SIP,agent-newlocation)
exten => 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})
Above two lines are very suspicious: AGENT_SIP is a variable which is
getting some DTMF from caller. agent-newlocation is the message you
want to be played while getting the AGENT_SIP input.
What is the next line doing in SET() !!? Please explain that.
Regards,
Sammy
On Wed, Jul 4, 2012 at 2:32 PM, Jakob-Matthias Böttger > wrote:
Hello
i got two Asterisk servers connected with IAX2.
At server 1 i hosted a queue named support. Now i want the Agent
to login from SIP Phones Connected to server B.
Therefore i wrote a extension like
exten => 105,1,Authenticate(1234)
exten => 105,n,AddQueueMember(support,,)
exten => 105,n,Read(AGENT_SIP,agent-newlocation)
exten => 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})
exten => 105,n,Playback(agent-loginok)
exten => 105,n,Playback(vm-goodbye)
exten => 105,n,Hangup
Now server A takes IAX/serverB as argument for AGENT_SIP but
deletes the entered number in agent-newlocation. Do you have any
ideas how to solve that?
Best regards Jakob
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Re: [asterisk-users] Queue Member login from IAX trunk
Hi,
exten => 105,n,Read(AGENT_SIP,agent-**newlocation)
exten => 105,n,Set(AGENT_SIP=${DB(**agent_sip/${agent-newlocation}**)})
Above two lines are very suspicious: AGENT_SIP is a variable which is
getting some DTMF from caller. agent-**newlocation is the message you
want to be played while getting the AGENT_SIP input.
What is the next line doing in SET() !!? Please explain that.
Regards,
Sammy
On Wed, Jul 4, 2012 at 2:32 PM, Jakob-Matthias Böttger wrote:
> Hello
>
> i got two Asterisk servers connected with IAX2.
>
> At server 1 i hosted a queue named support. Now i want the Agent to login
> from SIP Phones Connected to server B.
> Therefore i wrote a extension like
>
> exten => 105,1,Authenticate(1234)
> exten => 105,n,AddQueueMember(support,,**)
> exten => 105,n,Read(AGENT_SIP,agent-**newlocation)
> exten => 105,n,Set(AGENT_SIP=${DB(**agent_sip/${agent-newlocation}**)})
> exten => 105,n,Playback(agent-loginok)
> exten => 105,n,Playback(vm-goodbye)
> exten => 105,n,Hangup
>
> Now server A takes IAX/serverB as argument for AGENT_SIP but deletes the
> entered number in agent-newlocation. Do you have any ideas how to solve
> that?
>
> Best regards Jakob
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] port 5060 is blocked by ISP
Hi, Being audible sometime or bad voice quality is only due to internet latency or bad internet situation. [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt: Retransmission timeout reached on transmission 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up call 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). The above lines again telling that there is some problem sending sequential packets to some endpoint. That may lead to disconnection of call after some time..as it is currently doing so. Try setting some more NAT parameters...as you said localnet. Set *localnet=*parameter entries in your asterisk server sip configurations. BR Sammy On Wed, Jul 4, 2012 at 2:13 PM, alok srivastava wrote: > thanks Samy > i have set nat=yes, now getting sound from both side but there is too uch > disturbance. soetime we becoe audible and sometime not.i did not set extern > ip coz my asterisk server is directly configured on public ip. I have > softphones on some where localnets separate from asterisk server campus . i > also set "sip set debug on" CLI prompt. this is giving following error. > > when i test sip traffic on wireshark "401 unauthorize" error getting this > error cli prompt also showing. > > my first softph(9001) is on localnet 192.168.1.136 and 2nd softphone > (9000) in another localnet in another campus(192.168.6.25) > > > Scheduling destruction of SIP dialog ' > 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' in 32000 ms > (Method: INVITE) > [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt: > Retransmission timeout reached on transmission > 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 for seqno 102 > (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > Packet timed out after 32000ms with no response > [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up > call 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 - no reply to > our critical packet (see > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). > -- SIP/9000-0005 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > -- Auto fallthrough, channel 'SIP/9001-0004' status is 'CONGESTION' > > <--- Reliably Transmitting (NAT) to 122.163.193.94:1801 ---> > SIP/2.0 503 Service Unavailable > Via: SIP/2.0/UDP 192.168.1.136:5060 > ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;received=122.163.193.94;rport=1801 > From: "9001";tag=b0785362 > To: ;tag=as6c7d28d1 > Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. > CSeq: 2 INVITE > Server: Asterisk PBX 10.0.0 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > X-Asterisk-HangupCause: No user responding > X-Asterisk-HangupCauseCode: 18 > Content-Length: 0 > > > <> > Really destroying SIP dialog ' > 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' Method: INVITE > > <--- SIP read from UDP:122.163.193.94:1801 ---> > ACK sip:9000@122.160.154.189 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.136:5060 > ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;rport > Max-Forwards: 70 > To: ;tag=as6c7d28d1 > From: "9001";tag=b0785362 > Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. > CSeq: 2 ACK > Content-Length: 0 > > <-> > --- (8 headers 0 lines) --- > Really destroying SIP dialog > 'MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.' Method: ACK > > <--- SIP read from UDP:122.163.193.94:1801 ---> > > > <-> > > <--- SIP read from UDP:115.249.67.250:5060 ---> > REGISTER sip:122.160.154.189 SIP/2.0 > Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKskzxkdlp > Max-Forwards: 70 > To: "shekhar" > From: "shekhar" ;tag=jcysf > Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 > CSeq: 954 REGISTER > Contact: ;expires=3600 > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE > User-Agent: Twinkle/1.4.2 > Content-Length: 0 > > <-> > --- (11 headers 0 lines) --- > Sending to 115.249.67.250:5060 (NAT) > > <--- Transmitting (NAT) to 115.249.67.250:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 192.168.6.25;branch=z9hG4bKskzxkdlp;received=115.249.67.250;rport=5060 > From: "shekhar" ;tag=jcysf > To: "shekhar" ;tag=as26d4cd86 > Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 > CSeq: 954 REGISTER > Server: Asterisk PBX 10.0.0 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="278a3764" > Content-Length: 0 > > > <> > Scheduling destruction of SIP dialog 'qajoim
[asterisk-users] Queue Member login from IAX trunk
Hello
i got two Asterisk servers connected with IAX2.
At server 1 i hosted a queue named support. Now i want the Agent to
login from SIP Phones Connected to server B.
Therefore i wrote a extension like
exten => 105,1,Authenticate(1234)
exten => 105,n,AddQueueMember(support,,)
exten => 105,n,Read(AGENT_SIP,agent-newlocation)
exten => 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})
exten => 105,n,Playback(agent-loginok)
exten => 105,n,Playback(vm-goodbye)
exten => 105,n,Hangup
Now server A takes IAX/serverB as argument for AGENT_SIP but deletes the
entered number in agent-newlocation. Do you have any ideas how to solve
that?
Best regards Jakob
smime.p7s
Description: S/MIME Kryptografische Unterschrift
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Re: [asterisk-users] port 5060 is blocked by ISP
thanks Samy i have set nat=yes, now getting sound from both side but there is too uch disturbance. soetime we becoe audible and sometime not.i did not set extern ip coz my asterisk server is directly configured on public ip. I have softphones on some where localnets separate from asterisk server campus . i also set "sip set debug on" CLI prompt. this is giving following error. when i test sip traffic on wireshark "401 unauthorize" error getting this error cli prompt also showing. my first softph(9001) is on localnet 192.168.1.136 and 2nd softphone (9000) in another localnet in another campus(192.168.6.25) Scheduling destruction of SIP dialog ' 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' in 32000 ms (Method: INVITE) [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt: Retransmission timeout reached on transmission 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up call 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). -- SIP/9000-0005 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/9001-0004' status is 'CONGESTION' <--- Reliably Transmitting (NAT) to 122.163.193.94:1801 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;received=122.163.193.94;rport=1801 From: "9001";tag=b0785362 To: ;tag=as6c7d28d1 Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. CSeq: 2 INVITE Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0 <> Really destroying SIP dialog ' 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' Method: INVITE <--- SIP read from UDP:122.163.193.94:1801 ---> ACK sip:9000@122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;rport Max-Forwards: 70 To: ;tag=as6c7d28d1 From: "9001";tag=b0785362 Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. CSeq: 2 ACK Content-Length: 0 <-> --- (8 headers 0 lines) --- Really destroying SIP dialog 'MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.' Method: ACK <--- SIP read from UDP:122.163.193.94:1801 ---> <-> <--- SIP read from UDP:115.249.67.250:5060 ---> REGISTER sip:122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKskzxkdlp Max-Forwards: 70 To: "shekhar" From: "shekhar" ;tag=jcysf Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 954 REGISTER Contact: ;expires=3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 <-> --- (11 headers 0 lines) --- Sending to 115.249.67.250:5060 (NAT) <--- Transmitting (NAT) to 115.249.67.250:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKskzxkdlp;received=115.249.67.250;rport=5060 From: "shekhar" ;tag=jcysf To: "shekhar" ;tag=as26d4cd86 Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 954 REGISTER Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="278a3764" Content-Length: 0 <> Scheduling destruction of SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:115.249.67.250:5060 ---> REGISTER sip:122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKjmqlllxn Max-Forwards: 70 To: "shekhar" From: "shekhar" ;tag=jcysf Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 955 REGISTER Contact: ;expires=3600 Authorization: Digest username="9000",realm="asterisk",nonce="278a3764",uri="sip:122.160.154.189",response="c7a119185514202d5f9cc10a86a93607",algorithm=MD5 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 <-> --- (12 headers 0 lines) --- Sending to 115.249.67.250:5060 (NAT) <--- Transmitting (NAT) to 115.249.67.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKjmqlllxn;received=115.249.67.250;rport=5060 From: "shekhar" ;tag=jcysf To: "shekhar" ;tag=as26d4cd86 Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 955 REGISTER Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: ;expires=3600 Date: Wed, 04 Jul 2012 14:08:17 GMT Content-Length: 0 <> Scheduling destruct
