date:20111119

Re: [asterisk-users] SIP registration issues

2011-11-19 Thread Terry Wilson

I have not looked at the log files, but often times DSL routers may use PPPoE 
which has a little bit of overhead so you need to set the MTU below the default 
of 1500. Some info about the issue can be found here: 
http://www.ezlan.net/PPPOE.html and 
http://www.cisco.com/en/US/tech/tk175/tk15/technologies_tech_note09186a0080093bc7.shtml.

Another issue could be that the DSL router is doing a nat and you need to set 
nat=yes in sip.conf to get things to work.

- Original Message -
> From: "Raj Mathur (राज माथुर)" 
> To: asterisk-users@lists.digium.com
> Sent: Saturday, November 19, 2011 8:43:22 PM
> Subject: [asterisk-users] SIP registration issues
> Hi,
> 
> Having problems with a client trying to login to Asterisk 1.6.2 from
> behind a DSL router. The account can be accessed perfectly from other
> clients.
> 
> Would appreciate if you could look at the the attached log and see if
> you spot any glaring issues. The user is very infrequently available
> for discussion and testing, so please try to batch questions in one
> mail
> itself!
> 
> Regards,
> 
> -- Raj
> --
> Raj Mathur || r...@kandalaya.org || GPG:
> http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
> It is the mind that moves || http://schizoid.in || D17F
> 
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[asterisk-users] SIP registration issues

2011-11-19 Thread Raj Mathur (राज माथुर)

Hi,

Having problems with a client trying to login to Asterisk 1.6.2 from 
behind a DSL router.  The account can be accessed perfectly from other 
clients.

Would appreciate if you could look at the the attached log and see if
you spot any glaring issues.  The user is very infrequently available 
for discussion and testing, so please try to batch questions in one mail 
itself!

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F
REGISTER sip:SERVER-IP SIP/2.0
CSeq: 100 REGISTER
Via: SIP/2.0/UDP 
CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport
User-Agent: Ekiga/3.2.7
From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d
Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway
To: 
Contact: ;q=1, 
;q=0.667, 
;q=0.334
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Content-Length: 0
Max-Forwards: 70


<->
--- (12 headers 0 lines) ---
Sending to CLIENT-IP : 49153 (no NAT)

<--- Transmitting (no NAT) to CLIENT-IP:49153 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152
From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d
To: ;tag=as5d35c321
Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway
CSeq: 100 REGISTER
Server: Asterisk PBX 1.6.2.9-2+squeeze3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09c83283"
Content-Length: 0


<>
Scheduling destruction of SIP dialog 
'0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:CLIENT-IP:49152 --->
REGISTER sip:SERVER-IP SIP/2.0
CSeq: 100 REGISTER
Via: SIP/2.0/UDP 
CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport
User-Agent: Ekiga/3.2.7
From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d
Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway
To: 
Contact: ;q=1, 
;q=0.667, 
;q=0.334
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Content-Length: 0
Max-Forwards: 70


<->
--- (12 headers 0 lines) ---
Sending to CLIENT-IP : 49153 (no NAT)

<--- Transmitting (no NAT) to CLIENT-IP:49153 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152
From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d
To: ;tag=as5d35c321
Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway
CSeq: 100 REGISTER
Server: Asterisk PBX 1.6.2.9-2+squeeze3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09c83283"
Content-Length: 0


<>
Scheduling destruction of SIP dialog 
'0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:CLIENT-IP:49152 --->
REGISTER sip:SERVER-IP SIP/2.0
CSeq: 100 REGISTER
Via: SIP/2.0/UDP 
CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport
User-Agent: Ekiga/3.2.7
From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d
Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway
To: 
Contact: ;q=1, 
;q=0.667, 
;q=0.334
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Content-Length: 0
Max-Forwards: 70


<->
--- (12 headers 0 lines) ---
Sending to CLIENT-IP : 49153 (no NAT)

<--- Transmitting (no NAT) to CLIENT-IP:49153 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152
From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d
To: ;tag=as5d35c321
Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway
CSeq: 100 REGISTER
Server: Asterisk PBX 1.6.2.9-2+squeeze3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09c83283"
Content-Length: 0


<>
Scheduling destruction of SIP dialog 
'0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:CLIENT-IP:49152 --->
REGISTER sip:SERVER-IP SIP/2.0
CSeq: 100 REGISTER
Via: SIP/2.0/UDP 
CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport
User-Agent: Ekiga/3.2.7
From: ;tag=0d5779ba-a505-1910-8afe-001302871a3d
Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway
To: 
Contact: ;q=1, 
;q=0.667, 
;q=0.334
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Expires: 3600
Content-Length: 0
Max-Forwards: 70


<->
--- (12 headers 0 lines) ---
Sending to CLIENT-IP : 49153 (no NAT)

<--- Transmitting (no NAT) to CLIENT-IP:49153 --->
SIP/2.0 401 Unauthorized
V

Re: [asterisk-users] Question about Read() application

2011-11-19 Thread Kingsley Tart

Hi,

Did you get a workaround for this? I sent you a message offlist but you
didn't reply so I don't know whether you saw it.

Cheers,
Kingsley.

On Fri, 2011-11-18 at 13:15 -0600, Danny Nicholas wrote:
> My IVR wouldn't sound right if I allowed 2 or 3 times before it was
> considered a failure.   The big(ger) problem is that it just hangs up when
> it fails, no warning or work around to do.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
> Sent: Friday, November 18, 2011 1:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question about Read() application
> 
> 
> Danny Nicholas wrote:
> > The user reported to me that "I punched 1 and it hung up" - in my testing,
> I found that slow DTMF entry (1 digit every 2 seconds or so) or fast entry
> (more than 10 digits per second) was most likely to cause the problem.
> 
> I've never had mine just hangup on a mis-key, but then again I have it try 3
> times before considering it a failure.
> 
> exten => s,1,Read(get-admin-password|enter-password|||3|)
> 
> Doug
> 
> 


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[asterisk-users] Queue: The call keep going to agent until the agent drop the call

2011-11-19 Thread bilal ghayyad

Hi All;

I am facing a problem that when the call send to the queue, it is sending to 
the agent while the agent already has a call!! And it keeps sending the calls 
until the agent hangup the call the first call, then the calls go to the next 
agent and so on. And while the agent is having the call, he can see the waited 
calls at his IP Phone, until he hangup the first call and the waited call 
answered, so the next calls go for the next agent (and the same scenario 
repeated). WHY?!! How I can resolve this? Because when the agent answered the 
call, then new call should goes to the next agent and not goes to the same 
agent and appeared on his IP Phone, while other agents are idle !! Note: all 
agents have the same penalty.

I tried to different combinations of settings for autofill and ringinuse 
without any success.

Another thing I need to notify, that it is configured on the IP Phone only one 
extension.

Below is my extensions.conf and queues.conf settings:

extensions.conf

[OrangeCMG]

include => Internal

exten => 
s,1,Set(MONITOR_FILENAME=${CHANNEL}${CALLERID(num)}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten => s,2,Queue(OrangeCMG,t,,,300)
exten => s,3,Macro(voicemail,SIP/reception)
;

queues.conf

[OrangeCMG]

musicclass = default
announce = queue-markq
strategy = fewestcalls ; I also tried leastrecent
context = ExternalAgent
penaltymemberslimit = 5 ; I also tried penaltymemberslimit = 0
autofill = yes ; I also tried autofill = no
maxlen = 0
setinterfacevar=no ; I also tried setinterfacevar=yes
monitor-format = wav
monitor-type = MixMonitor
ringinuse = no ; I also tried ringinuse = yes


Regards
Bilal

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Re: [asterisk-users] SIP registration issues

[asterisk-users] SIP registration issues

Re: [asterisk-users] Question about Read() application

[asterisk-users] Queue: The call keep going to agent until the agent drop the call

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