Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Julian Beach]

Hello Phil,

On Saturday, April 23, 2016, 11:11:29 PM, you wrote:

> Actually, this is now sorted. It turns out the latest recommended
> configs on the A wiki had peer vs. user confusion. On correcting
> this, all was well.

I'm glad you found it. It look me a while to track down that problem
when I had it.

The one that was hardest for me to track down was a slight mis-match
between the RTP ports in Asterisk and the corresponding ports open on
a firewall, which resulted in about 1 in 10 calls having no audio!
Doh!

-- 
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 Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Phil Reynolds]

On Sat, 23 Apr 2016 22:45:32 +0100
Julian Beach  wrote:

> Hello Phil,
> 
> I have a couple of lines with A, and I have not been having any
> problems recently. When I have had similar problems in the past, it
> has been an issue with the SIP config. I originally had a number of
> contexts set up in sip.conf to handle the lines coming in (such as
> [aa-line1], [aa-line2]) each with their own username and password
> settings. The type=user setting was critical, because all the calls
> came from the same IP address, and using type=peer caused matching
> problems which resulted in authentication failures. This got too
> complex to manage once I added in all the IP addresses A calls might
> come in from. so I simplified the setup.
> 
> I now have just one context in sip.conf to handle incoming A calls,
> with the same username for all lines, and type=peer. Calls are then
> sent to extensions.conf, where the calls are directed to the correct
> call-handler for the line based on the CID. Here is the setup in
> sip.conf for A calls:

Actually, this is now sorted. It turns out the latest recommended
configs on the A wiki had peer vs. user confusion. On correcting
this, all was well.

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/

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Re: [asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Julian Beach]

Hello Phil,

On Saturday, April 23, 2016, 12:19:15 PM, you wrote:

> I have checked that the username and password in my config agree both
> ends, and have even tried changing them.

> The bulk of my calls come in on A, so I am obviously trying to find
> out what has gone wrong. No-one else is seeing any problem. What do I
> need to do to track this down?

I have a couple of lines with A, and I have not been having any
problems recently. When I have had similar problems in the past, it
has been an issue with the SIP config. I originally had a number of
contexts set up in sip.conf to handle the lines coming in (such as
[aa-line1], [aa-line2]) each with their own username and password
settings. The type=user setting was critical, because all the calls
came from the same IP address, and using type=peer caused matching
problems which resulted in authentication failures. This got too
complex to manage once I added in all the IP addresses A calls might
come in from. so I simplified the setup.

I now have just one context in sip.conf to handle incoming A calls,
with the same username for all lines, and type=peer. Calls are then
sent to extensions.conf, where the calls are directed to the correct
call-handler for the line based on the CID. Here is the setup in
sip.conf for A calls:

---
sip.conf

[aa-incoming](!)
type=peer
context=aa-incoming
insecure=invite
transport=udp
disallow=all
allow=alaw
trustrpid=yes
sendrpid=yes

; IPv4 hostnames
[voiceless-1](aa-incoming)
host=a4.voiceless.aa.net.uk
[voiceless-2](aa-incoming)
host=b4.voiceless.aa.net.uk
[voiceless-3](aa-incoming)
host=c4.voiceless.aa.net.uk
[voiceless-4](aa-incoming)
host=d4.voiceless.aa.net.uk
[voiceless-5](aa-incoming)
host=e4.voiceless.aa.net.uk
[voiceless-6](aa-incoming)
host=f4.voiceless.aa.net.uk
[voiceless-7](aa-incoming)
host=g4.voiceless.aa.net.uk
[voiceless-8](aa-incoming)
host=h4.voiceless.aa.net.uk
[voiceless-9](aa-incoming)
host=i4.voiceless.aa.net.uk
[voiceless-10](aa-incoming)
host=j4.voiceless.aa.net.uk
---

The trustrpid and sendrpid settings were important.

---
extensions.conf  (DNIDs changed)
===
[aa-incoming]
exten => 4401,1,Goto(from-aa-line1,s,1)
exten => 4402,1,Goto(from-aa-line2,s,1)
exten => 4403,1,Goto(from-aa-line3,s,1)
---

Hope this helps.

Julian





-- 
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 Julianmailto:jb_s...@trink.co.uk


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[asterisk-users] Incoming calls from Andrews & Arnold failing to authenticate

[Phil Reynolds]

I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk).
VoIPtalk calls are unauthenticated and reach me fine, but Andrews &
Arnold calls are authenticated. The last call I successfully received
was on Tuesday afternoon. Initially, A were for some odd reason not
sending calls to my server, but that has been resolved. The problem now
is that the calls fail to authenticate, and are therefore rejected -
error 403 is presented to them, and I see this in Asterisk's console: 

[Apr 23 11:53:19] NOTICE[27398][C-0004]: chan_sip.c:25535
handle_request_invite: Failed to authenticate device "X XX"
;tag=201604231153191

I have checked that the username and password in my config agree both
ends, and have even tried changing them.

The bulk of my calls come in on A, so I am obviously trying to find
out what has gone wrong. No-one else is seeing any problem. What do I
need to do to track this down?

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/

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Re: [asterisk-users] Incoming calls get 488 error

[Andres]

On 8/21/15 6:45 PM, Technical Support wrote:
I got a new SNOM M65 which works fine for outgoing calls, but incoming 
calls never ring at the handset.  I captured the SIP traffic and see 
that my M65 is replying with an 488 not acceptable here. From what I 
read this is usually codec related but both asterisk and the M65 are 
set for ulaw as first choice.
Looks like the SNOM does not accept the video call.  Maybe you should 
look into why the Asterisk is trying to use video in the first place.


I have a SIP trace below.  Can someone suggest why the 488 is being 
generated?


---

Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (1198 bytes)

INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Contact: sip:230@192.168.253.4:5060
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606

v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/9
a=fmtp:99 
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0

a=rtpmap:98 H263-1998/9
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/9
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/9
a=sendrecv


Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (280 bytes)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0



Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (441 bytes)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: sip:290006@192.168.253.20;line=14994
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255)
Content-Length: 0






--
Technical Support
http://www.cellroute.net


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[asterisk-users] Incoming calls get 488 error

[Technical Support]

I got a new SNOM M65 which works fine for outgoing calls, but incoming 
calls never ring at the handset.  I captured the SIP traffic and see 
that my M65 is replying with an 488 not acceptable here.  From what I 
read this is usually codec related but both asterisk and the M65 are set 
for ulaw as first choice.


I have a SIP trace below.  Can someone suggest why the 488 is being 
generated?


---

Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (1198 bytes)

INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Contact: sip:230@192.168.253.4:5060
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606

v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/9
a=fmtp:99 
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0

a=rtpmap:98 H263-1998/9
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/9
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/9
a=sendrecv


Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (280 bytes)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0



Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (441 bytes)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: sip:290006@192.168.253.20;line=14994
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255)
Content-Length: 0



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Re: [asterisk-users] Incoming calls get 488 error

[Rafael Prado Rocchi]

Hi,
By the sip trace is very difficult to tell because the SIP messages are fine. 
Try to enable all codec, and if possible copy and paste your asterisk sip 
configuration for this peer.



Enviado do meu telefone Android usando o Symantec TouchDown (www.symantec.com)


-Original Message-
From: Technical Support [supp...@telium.ca]
Received: sexta-feira, 21 ago 2015, 19:46
To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com]
Subject: [asterisk-users] Incoming calls get 488 error


I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset.  I captured the SIP traffic and see
that my M65 is replying with an 488 not acceptable here.  From what I
read this is usually codec related but both asterisk and the M65 are set
for ulaw as first choice.

I have a SIP trace below.  Can someone suggest why the 488 is being
generated?

---

Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (1198 bytes)

INVITE sip:290006@192.168.253.20;line=14994 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Contact: sip:230@192.168.253.4:5060
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.10.2)
Date: Fri, 21 Aug 2015 22:37:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 606

v=0
o=root 1678280845 1678280845 IN IP4 192.168.253.4
s=Asterisk PBX 11.10.2
c=IN IP4 192.168.253.4
b=CT:384
t=0 0
m=audio 18090 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12226 RTP/AVP 99 98 34 31
a=rtpmap:99 H264/9
a=fmtp:99
redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/9
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/9
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/9
a=sendrecv


Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (280 bytes)

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Content-Length: 0



Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00  (441 bytes)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
Max-Forwards: 70
From: test user sip:230@192.168.253.4;tag=as7b616c8d
To: sip:290006@192.168.253.20;line=14994;tag=ld65q
Call-ID: 36334383058109cd2325341a0f18ac79@192.168.253.4:5060
CSeq: 102 INVITE
Contact: sip:290006@192.168.253.20;line=14994
User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 0; HW=255)
Content-Length: 0



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[asterisk-users] Incoming calls to a GSM gateway SIP/2.0 401 Unauthorized response when dial 7777 to Asterisk

[Luis Eduardo Cortes]

Hello:

I'm newbie in asterisk, please help me.

My context is as follows:

192.168.4.2 -- Asterisk 11.13.1 complied from source

192.168.4.4 -- Yeastar NeoGate TG100 GSM gateway

When I call from a GSM cell phone, my TG100 GSM gateway answers and
dials extension  (configured as a hotline on TG100) to asterisk
server, but asterisk server sends me SIP/2.0 401 Unauthorized
response, I think it's a matter of contexts but I don't find the
problem.

Attached are sip.conf, extensions.conf and debug from 192.168.4.4
(TG100 GSM gateway).

Thanks in advance.
--- SIP read from UDP:192.168.4.4:5060 ---
INVITE sip:@192.168.4.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport
Max-Forwards: 70
From: 9 sip:9@192.168.4.4;tag=as67354416
To: sip:@192.168.4.2:5060
Contact: sip:9@192.168.4.4
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 INVITE
User-Agent: TG100
Date: Wed, 12 Nov 2014 10:13:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1426707418 1426707418 IN IP4 192.168.4.4
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.4.4
t=0 0
m=audio 10048 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-
--- (14 headers 13 lines) ---
Sending to 192.168.4.4:5060 (no NAT)
Sending to 192.168.4.4:5060 (no NAT)
Using INVITE request as basis request - 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
Found peer '5' for '9' from 192.168.4.4:5060

--- Reliably Transmitting (no NAT) to 192.168.4.4:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;received=192.168.4.4;rport=5060
From: 9 sip:9@192.168.4.4;tag=as67354416
To: sip:@192.168.4.2:5060;tag=as16de6e5c
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=72011a6b
Content-Length: 0



Scheduling destruction of SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' in 6400 ms (Method: INVITE)

--- SIP read from UDP:192.168.4.4:5060 ---
ACK sip:@192.168.4.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport
Max-Forwards: 70
From: 9 sip:9@192.168.4.4;tag=as67354416
To: sip:@192.168.4.2:5060;tag=as16de6e5c
Contact: sip:9@192.168.4.4
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 ACK
User-Agent: TG100
Content-Length: 0

-
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6e9eab843b74d0860b108ed13a5d22c9@192.168.4.4' Method: OPTIONS
Really destroying SIP dialog '12663beb04ae514f10c4b3a145368d5c@192.168.4.4' Method: ACK
uc*CLI 



extensions.conf
Description: Binary data


sip.conf
Description: Binary data
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Re: [asterisk-users] incoming calls fall into echo test mode

[A J Stiles]

On Saturday 19 Jul 2014, Norman Molhant wrote:

 I tried many things on our FreePBX box and found out
 the problem seems somehow linked with the customer's
 extension (or phone number), not his inbound route
 (changing the latter has no effect on the problem).
 
 Creating a new extension with another phone number
 would solve the problem (I tried it and it works),
 but this customer wants to keep his current phone
 number and when I tried deleting his extension then
 creating a new one with his current phone number,
 the new extension presented the same problem as the
 previous one...
 
 Anyone knows what could cause such a problem and/or
 how to solve it ?

You really have supplied incomplete information here, by neglecting to mention 
the actual extension number which is causing the problems.  That would have 
had somebody onto it like a shot.  What follows is an educated guess based on 
the most likely scenario according to the available information:

Somewhere in your dialplan, probably in a section that has already been 
helpfully configured for you by FreePBX, the extension number you assigned to 
your customer has been appropriated for an echotest.

I suggest to grep for  (firstly)  the extension number in question, and  (if 
that does not work, perhaps because the echotest is a wildcard match aot a 
literal one)  then search instead for 'exten[ ]*='  (afraid that one will 
give you many more hits .  you'll have to look through them yourself)  
under /etc/asterisk.  Use the -R option to search subfolders as well.


-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] incoming calls fall into echo test mode

[Norman Molhant]

Hello all,

Weird trouble here:
we have 60-some happy subscribers on a FreePBX box,
each with its own phone number, with no problem at all,
except for one (and only one) subscriber who has this
problem: his outgoing calls are ok, but when someone
dials his phone number (be it from our network or from
any other place in the world), the caller ears the
standard message signalling he has entered the echo
test mode and must dial # to exit that mode.

Most callers don't understand what's going on, then
give up and hang up without dialling #.  Very few
dial # one or more times, then those few get our
customer's phone ringing and are then able to reach
our customer.

I went through all the docs, wikis and discussions
I found on the web, without finding any data on how
to solve that problem.

I tried many things on our FreePBX box and found out
the problem seems somehow linked with the customer's
extension (or phone number), not his inbound route
(changing the latter has no effect on the problem).

Creating a new extension with another phone number
would solve the problem (I tried it and it works),
but this customer wants to keep his current phone
number and when I tried deleting his extension then
creating a new one with his current phone number,
the new extension presented the same problem as the
previous one...

Anyone knows what could cause such a problem and/or
how to solve it ?

Thanks,
Norman.
ad...@csur.ca






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Re: [asterisk-users] incoming calls fall into echo test mode

[Michelle Dupuis]

You might get a better response on the FreePBX forum.  (FreePBX adds pre-built 
dialplan elements onto standard asterisk.  This forum is more for Asterisk)

But some suggestions:

SSH to your PBX
enter the Asterisk CLI
set verbose to 10
Call into the problematic number
...and watch where the call is being misrouted in the dialplan



From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Norman Molhant 
ad...@csur.ca
Sent: Saturday, July 19, 2014 10:43 AM
To: Asterisk Users List
Subject: [asterisk-users] incoming calls fall into echo test mode

Hello all,

Weird trouble here:
we have 60-some happy subscribers on a FreePBX box,
each with its own phone number, with no problem at all,
except for one (and only one) subscriber who has this
problem: his outgoing calls are ok, but when someone
dials his phone number (be it from our network or from
any other place in the world), the caller ears the
standard message signalling he has entered the echo
test mode and must dial # to exit that mode.

Most callers don't understand what's going on, then
give up and hang up without dialling #.  Very few
dial # one or more times, then those few get our
customer's phone ringing and are then able to reach
our customer.

I went through all the docs, wikis and discussions
I found on the web, without finding any data on how
to solve that problem.

I tried many things on our FreePBX box and found out
the problem seems somehow linked with the customer's
extension (or phone number), not his inbound route
(changing the latter has no effect on the problem).

Creating a new extension with another phone number
would solve the problem (I tried it and it works),
but this customer wants to keep his current phone
number and when I tried deleting his extension then
creating a new one with his current phone number,
the new extension presented the same problem as the
previous one...

Anyone knows what could cause such a problem and/or
how to solve it ?

Thanks,
Norman.
ad...@csur.ca






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Re: [asterisk-users] incoming calls fall into echo test mode

[Pat Collins]

Perhaps assigned as a test number somewhere along the line?
Are these ISDN, SIP, IAX calls?
There are MANY smart people on this list. 
Maybe sharing the relevant configs and traces is a good place to start???

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant
Sent: Saturday, July 19, 2014 10:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming calls fall into echo test mode

Hello all,

Weird trouble here:
we have 60-some happy subscribers on a FreePBX box, each with its own phone
number, with no problem at all, except for one (and only one) subscriber who
has this
problem: his outgoing calls are ok, but when someone dials his phone number
(be it from our network or from any other place in the world), the caller
ears the standard message signalling he has entered the echo test mode and
must dial # to exit that mode.

Most callers don't understand what's going on, then give up and hang up
without dialling #.  Very few dial # one or more times, then those few get
our customer's phone ringing and are then able to reach our customer.

I went through all the docs, wikis and discussions I found on the web,
without finding any data on how to solve that problem.

I tried many things on our FreePBX box and found out the problem seems
somehow linked with the customer's extension (or phone number), not his
inbound route (changing the latter has no effect on the problem).

Creating a new extension with another phone number would solve the problem
(I tried it and it works), but this customer wants to keep his current phone
number and when I tried deleting his extension then creating a new one with
his current phone number, the new extension presented the same problem as
the previous one...

Anyone knows what could cause such a problem and/or how to solve it ?

Thanks,
Norman.
ad...@csur.ca






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Re: [asterisk-users] incoming calls fall into echo test mode

[covici]

check your logs /var/log/asterisk/full -- make sure your verbosity is
set high enough to do you good and you wll probably find the answer.

Pat Collins drdialt...@optonline.net wrote:

 Perhaps assigned as a test number somewhere along the line?
 Are these ISDN, SIP, IAX calls?
 There are MANY smart people on this list. 
 Maybe sharing the relevant configs and traces is a good place to start???
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant
 Sent: Saturday, July 19, 2014 10:43 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] incoming calls fall into echo test mode
 
 Hello all,
 
 Weird trouble here:
 we have 60-some happy subscribers on a FreePBX box, each with its own phone
 number, with no problem at all, except for one (and only one) subscriber who
 has this
 problem: his outgoing calls are ok, but when someone dials his phone number
 (be it from our network or from any other place in the world), the caller
 ears the standard message signalling he has entered the echo test mode and
 must dial # to exit that mode.
 
 Most callers don't understand what's going on, then give up and hang up
 without dialling #.  Very few dial # one or more times, then those few get
 our customer's phone ringing and are then able to reach our customer.
 
 I went through all the docs, wikis and discussions I found on the web,
 without finding any data on how to solve that problem.
 
 I tried many things on our FreePBX box and found out the problem seems
 somehow linked with the customer's extension (or phone number), not his
 inbound route (changing the latter has no effect on the problem).
 
 Creating a new extension with another phone number would solve the problem
 (I tried it and it works), but this customer wants to keep his current phone
 number and when I tried deleting his extension then creating a new one with
 his current phone number, the new extension presented the same problem as
 the previous one...
 
 Anyone knows what could cause such a problem and/or how to solve it ?
 
 Thanks,
 Norman.
 ad...@csur.ca
 
 
 
 
 
 
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-- 
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How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

[Matt Watson]

Just out of curiosity, what country are you in?

I agree with the others in this thread, this seems very bizzare that the
telco requires you to do SS7 for dialup connections.  I would ask them for
specifics about the legal issues with what you are doing - it sounds to me
like they are just trying to upsell you on a more expensive product.

I am in Canada and we run exactly the configuration you are currently
doing... we still have dialup internet customers that dial into AS5300's via
PRI's.  Our telco has a PRI product gear specifically for this use... they
call it 'ISP-PRI' I'm not entirely sure what the restriction is on it I
have also just kind of assumed that it is inbound calls only, but I've never
tried making outbound calls on them.  I do know they 25-30% cheaper than our
regular voice PRIs though.

--
Matt


2010/11/24 José Pablo Méndez Soto aux...@gmail.com

 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.


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Re: [asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

[Robert Thomas]

Matt,

We are located on Costa Rica and so far there's just 1 TELCO running the
industrym with the CAFTA treatment the carrier had to open for
interconnection but they get to define the ground rules for the
interconnection.

They are arguing ISDN is and end customer circuit and you cannot use it to
resell service. You need an Interconnect circuit, to resell dial up
access, and they require you to support SS7 for Interconnect T1s

It's just the carrier raising the bar waiting for 90% of the competition to
drop off, and a little porcentage to make it.

We asked for quotes for multiple solutions, most of them ranging 15 000$,
and finally come to the conclusion Asterisk and SS7 would be the only viable
option for this project.

Cheers,

On Tue, Nov 30, 2010 at 9:21 AM, Matt Watson m...@mattgwatson.ca wrote:

 Just out of curiosity, what country are you in?

 I agree with the others in this thread, this seems very bizzare that the
 telco requires you to do SS7 for dialup connections.  I would ask them for
 specifics about the legal issues with what you are doing - it sounds to me
 like they are just trying to upsell you on a more expensive product.

 I am in Canada and we run exactly the configuration you are currently
 doing... we still have dialup internet customers that dial into AS5300's via
 PRI's.  Our telco has a PRI product gear specifically for this use... they
 call it 'ISP-PRI' I'm not entirely sure what the restriction is on it I
 have also just kind of assumed that it is inbound calls only, but I've never
 tried making outbound calls on them.  I do know they 25-30% cheaper than our
 regular voice PRIs though.

 --
 Matt


 2010/11/24 José Pablo Méndez Soto aux...@gmail.com

 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Robert
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[asterisk-users] Incoming calls through SS7 for data modem transmissions - possible??

[José Pablo Méndez Soto]

Hello,

We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.

The Telco they were buying the trunks to discovered this configuration and
restricted them due to legal conventions, and stated that in order to
continue doing this, they would have to talk SS7 directly.

We are planning on solving this by placing an Asterisk server with some
TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
AS5300 for the dial-up to complete after authenticating against a RADIUS
server.

My questions is: can we use only Asterisk to complete/terminate the dial-up
connection, removing the AS5300 out of the picture?

Current topology to be set-up:
Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

Ideal topology:
Telco -- SS7 -- TE410P-AsteriskServer -- Internet


Some posts talk about zapRAS being able to accomplish this, not quite sure
though

Sounds like possible:
http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com

Sounds like not possible:
http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


Thanks in advance,


*José Pablo Méndez
  *
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Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

[Cary Fitch]

I am not sure where you are and what legal conventions are involved.

 

Are you saying the Telco (and legal restrictions) say you can’t send calls
to the internet via the AS5300 but you can if Asterisk does it directly?
What is the “logic” in that?

 

Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? 

 

Or are you concerned about Asterisk handling the TDM to IP conversion in an
adequate manner?

 

I am not sure/aware myself that Asterisk will do a modem to IP conversion.
I think in your example the AS5300 is doing that.

 

What is the Telco’s problem in doing what the customer was doing before?

 

Feel free to correspond directly if you want to.

 

Cary Fitch

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo
Méndez Soto
Sent: Wednesday, November 24, 2010 7:31 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming calls through SS7 for data
modemtransmissions - possible??

 

Hello,

We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.

The Telco they were buying the trunks to discovered this configuration and
restricted them due to legal conventions, and stated that in order to
continue doing this, they would have to talk SS7 directly.

We are planning on solving this by placing an Asterisk server with some
TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
AS5300 for the dial-up to complete after authenticating against a RADIUS
server.

My questions is: can we use only Asterisk to complete/terminate the dial-up
connection, removing the AS5300 out of the picture?

Current topology to be set-up:
Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

Ideal topology:
Telco -- SS7 -- TE410P-AsteriskServer -- Internet


Some posts talk about zapRAS being able to accomplish this, not quite sure
though

Sounds like possible:
http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html
mailto:asterisk-users@lists.digium.com 

Sounds like not possible:
http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


Thanks in advance,


José Pablo Méndez
  

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Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??

[José Pablo Méndez Soto]

Thanks Cary,

What happens is, the Telco won't allow the small company to resell the ISDN
connections, meaning, they bought the trunks and DIDs, then sold dialing
plans to route incoming calls through the PRIs out the Internet. This is not
the issue though. We definitely have to migrate to an SS7 capable platform,
because that is the only way the Telco allows them to resell the dial-up
connections (not ISDN), and Asterisk is the current bet.

If we can get Asterisk to pick up those calls via SS7, then authenticate
them, send them out to the Internet, we would be achieving a %100 usage on
the Digium cards, because one of them wouldn't be used to talk to the AS.

Can Asterisk do this?


Thanks again,

*José Pablo Méndez
   *


On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch ca...@usawide.net wrote:

  I am not sure where you are and what legal conventions are involved.



 Are you saying the Telco (and legal restrictions) say you can’t send calls
 to the internet via the AS5300 but you can if Asterisk does it directly?
 What is the “logic” in that?



 Or are they saying your Telco to Asterisk trunks have to be SS7 controlled?




 Or are you concerned about Asterisk handling the TDM to IP conversion in an
 adequate manner?



 I am not sure/aware myself that Asterisk will do a modem to IP conversion.
 I think in your example the AS5300 is doing that.



 What is the Telco’s problem in doing what the customer was doing before?



 Feel free to correspond directly if you want to.



 Cary Fitch


  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez
 Soto
 *Sent:* Wednesday, November 24, 2010 7:31 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Incoming calls through SS7 for data
 modemtransmissions - possible??



 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.

 We are planning on solving this by placing an Asterisk server with some
 TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
 AS5300 for the dial-up to complete after authenticating against a RADIUS
 server.

 My questions is: can we use only Asterisk to complete/terminate the dial-up
 connection, removing the AS5300 out of the picture?

 Current topology to be set-up:
 Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

 Ideal topology:
 Telco -- SS7 -- TE410P-AsteriskServer -- Internet


 Some posts talk about zapRAS being able to accomplish this, not quite sure
 though

 Sounds like possible:
 http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com

 Sounds like not possible:
 http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


 Thanks in advance,


 *José Pablo Méndez**
   *

 --
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Re: [asterisk-users] Incoming calls through SS7 for datamodemtransmissions - possible??

[José Pablo Méndez Soto]

Thanks Cary,

The first topology we are working on should be the best way then.

Asterisk will answer SS7 calls, route them to the ISDN channels to be
terminated by the AS5300 as they were doing before. I think TDM-2-TDM
shouldn't be that much of a problem eh? No further equipment needed?


*José Pablo Méndez
   *


2010/11/24 Cary Fitch ca...@usawide.net

  I understand the problem.  You can’t resell PRI connections.



 I don’t think Asterisk can convert TDM to IP.  It does convert TDM to SIP
 which is then sent out over IP.What you want to do is have it do the
 TDM/Modem conversion without the PRIs and Cisco Gear.



 There used to be a way to do this, and maybe still is but not just with
 Asterisk perhaps.



 I know that Ascend/Lucent TNTs (and I am sure some other equipment)  could
 take TDM trunks, which could be SS7 trunks, and convert them to IP.



 The point in this is that they are SS7 based.  You can take SS7 trunks from
 either the Asterisk box or direct from the Telco and convert them to IP.



 NO PRIs involved.  Yes, more “telco grade carrier equipment” but no PRIs.



 A lot of this equipment was available by the pound a few years back.



 Cary


  --

 *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com]
 *Sent:* Wednesday, November 24, 2010 8:34 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Cc:* ca...@usawide.net
 *Subject:* Re: [asterisk-users] Incoming calls through SS7 for
 datamodemtransmissions - possible??



 Thanks Cary,

 What happens is, the Telco won't allow the small company to resell the ISDN
 connections, meaning, they bought the trunks and DIDs, then sold dialing
 plans to route incoming calls through the PRIs out the Internet. This is not
 the issue though. We definitely have to migrate to an SS7 capable platform,
 because that is the only way the Telco allows them to resell the dial-up
 connections (not ISDN), and Asterisk is the current bet.

 If we can get Asterisk to pick up those calls via SS7, then authenticate
 them, send them out to the Internet, we would be achieving a %100 usage on
 the Digium cards, because one of them wouldn't be used to talk to the AS.

 Can Asterisk do this?


 Thanks again,

 *José Pablo Méndez**
*

  On Wed, Nov 24, 2010 at 7:59 PM, Cary Fitch ca...@usawide.net wrote:

 I am not sure where you are and what legal conventions are involved.



 Are you saying the Telco (and legal restrictions) say you can’t send calls
 to the internet via the AS5300 but you can if Asterisk does it directly?
 What is the “logic” in that?



 Or are they saying your Telco to Asterisk trunks have to be SS7 controlled?




 Or are you concerned about Asterisk handling the TDM to IP conversion in an
 adequate manner?



 I am not sure/aware myself that Asterisk will do a modem to IP conversion.
 I think in your example the AS5300 is doing that.



 What is the Telco’s problem in doing what the customer was doing before?



 Feel free to correspond directly if you want to.



 Cary Fitch


  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez
 Soto
 *Sent:* Wednesday, November 24, 2010 7:31 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Incoming calls through SS7 for data
 modemtransmissions - possible??



 Hello,

 We are working on implementing a solution for a medium service provider.
 They were previously using a Cisco AS5300 gateway with some PRI trunks to
 receive modem calls, then route them out the Internet.

 The Telco they were buying the trunks to discovered this configuration and
 restricted them due to legal conventions, and stated that in order to
 continue doing this, they would have to talk SS7 directly.

 We are planning on solving this by placing an Asterisk server with some
 TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the
 AS5300 for the dial-up to complete after authenticating against a RADIUS
 server.

 My questions is: can we use only Asterisk to complete/terminate the dial-up
 connection, removing the AS5300 out of the picture?

 Current topology to be set-up:
 Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet

 Ideal topology:
 Telco -- SS7 -- TE410P-AsteriskServer -- Internet


 Some posts talk about zapRAS being able to accomplish this, not quite sure
 though

 Sounds like possible:
 http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.htmlasterisk-users@lists.digium.com

 Sounds like not possible:
 http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html


 Thanks in advance,


 *José Pablo Méndez**
   *


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[asterisk-users] Incoming calls

[Flavio Miranda]

Hi all,
  I'd like that each analog trunk of my TDM410p was received in different 
extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a 
different context and in my extensions.conf, under [default] I put such 
contexts and an especific estension to answer it. therefore, when I get  call, 
it always is ringing on the first extensions, dont matter trunk  . Anybody 
could teach me how can I organize that ?
 Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

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Re: [asterisk-users] Incoming calls

[Steve Edwards]

On Thu, 18 Nov 2010, Flavio Miranda wrote:

 I'd like that each analog trunk of my TDM410p was received in different 
 extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each 
 trunk in a different context and in my extensions.conf, under [default] 
 I put such contexts and an especific estension to answer it. therefore, 
 when I get call, it always is ringing on the first extensions, dont 
 matter trunk. Anybody could teach me how can I organize that ?

0) Use a subject that gives a clue what you're looking for. Almost 
everybody has had a question about an incomig call at some point in time.
Better bait = better fish.

1) It sounds like you have a clue about how to do it and are on the right 
track.

2) Including some details like the console output from:

zap show channel 1 (I'm a 1.2 Luddite.)
zap show channel 2
zap show channel 3
zap show channel 4

as well as the console log from a call coming in on each channel

will help in assisting you in resolving this issue.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Incoming calls

[Flavio Miranda]

Hi Steve,
thanks for the tips Better bait = better fish !

As you said, I  am in the right track.
Looking to dahdi show channles , I realized  that all the trunks was in the 
same context. So, I have changed  this and everything works!
thanks you !!
Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda



 Date: Thu, 18 Nov 2010 11:53:26 -0800
 From: asterisk@sedwards.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Incoming calls
 
 On Thu, 18 Nov 2010, Flavio Miranda wrote:
 
  I'd like that each analog trunk of my TDM410p was received in different 
  extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each 
  trunk in a different context and in my extensions.conf, under [default] 
  I put such contexts and an especific estension to answer it. therefore, 
  when I get call, it always is ringing on the first extensions, dont 
  matter trunk. Anybody could teach me how can I organize that ?
 
 0) Use a subject that gives a clue what you're looking for. Almost 
 everybody has had a question about an incomig call at some point in time.
 Better bait = better fish.
 
 1) It sounds like you have a clue about how to do it and are on the right 
 track.
 
 2) Including some details like the console output from:
 
 zap show channel 1 (I'm a 1.2 Luddite.)
 zap show channel 2
 zap show channel 3
 zap show channel 4
 
 as well as the console log from a call coming in on each channel
 
 will help in assisting you in resolving this issue.
 
 -- 
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
 
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Re: [asterisk-users] Incoming calls

[Steve Edwards]

On Thu, 18 Nov 2010, Flavio Miranda wrote:

Looking to dahdi show channles , I realized  that all the trunks was in 
the same context. So, I have changed  this and everything works!


That's why I prefer to work from what Asterisk parsed the file as, not 
what the poster thinks :)


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-
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[asterisk-users] Incoming calls

[Flavio Miranda]

Hi all,

   After a lot of trouble with a TE110p working with mfcr2 , brazil variant, 
everything looks great,but I can not go out of my calls.
When I try I receive the following  log:
== Using SIP RTP CoS mark 5-- Executing [33220...@local:1] 
Dial(SIP/4804-001a, DAHDI/g11/33220567,,T) in new stack  == Everyone is 
busy/congested at this time (1:0/1/0)-- Auto fallthrough, channel 
'SIP/4804-001a' status is 
'CONGESTION'This
 is  my dahdi show status:Digium Wildcard TE110P T1/E1 Card 0  REC 0
  0  0  CAS HDB3 CRC4 0 db (CSU)/0-133 feet 
(DSX-1)thi´s
 my dahdi show channels:
asterisk*CLI dahdi show channels   Chan Extension  Context Language   
MOH InterpretBlockedState  pseudodefault
default In Service  1 4800   default
default In Service  2 4800   
defaultdefault In Service  3 
4805   defaultdefault In 
Service  4defaultdefault
 In Service  5defaultdefault
 In Service  6defaultdefault
 In Service  7default
default In Service  8default
default In Service  9default
default In Service 10
defaultdefault In Service 11
defaultdefault In Service   
  12defaultdefault In 
Service 13defaultdefault
 In Service 14defaultdefault
 In Service 15defaultdefault
 In Service 17default
default In Service 18default
default In Service 19default
default In Service 20
defaultdefault In Service 21
defaultdefault In Service   
  22defaultdefault In 
Service 23defaultdefault
 In Service 24defaultdefault
 In Service 25defaultdefault
 In Service 26default
default In Service 27default
default In Service 28default
default In Service 29
defaultdefault In Service 30
defaultdefault In Service   
  31defaultdefault In 
Service
*In
 my incoming call , the log is:
MFC/R2 call offered on chan 1. ANI = 1221341400, DNIS = 4804, Category = 
National SubscriberNew MFC/R2 call detected on chan 2.

and  don't ring nowhere!
Thanks for help!Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

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[asterisk-users] Incoming calls coming into default context

[jonas kellens]

My SIP-provider sends my a SIP-invite like this :

INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0
Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
Max-Forwards: 70
From: sip:321445x...@80.xx.xx.69;tag=f395877e02bf8eb2fd8f5a0e
To: sip:329298y...@80.xx.xx.69
Call-ID: f395877e02187250fd8f5...@80.xx.xx.68
CSeq: 1 INVITE
User-Agent: SysMaster VoIP Gateway v1.2.0
Contact: sip:321445x...@80.xx.xx.68:5060
Remote-Party-ID:
sip:321445x...@80.xx.xx.69;party=calling;screen=yes;privacy=off
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 261

This is my sip.conf :

[outgoing]
type=peer
host=sip.XXX.tld
username=329298xxx6
secret=my-secret
fromuser=329298xxx6
disallow=all
allow=gsm
allow=alaw

; incoming
[329298yyy6]
type=user
host=sip.XXX.tld
context=mycontext
disallow=all
allow=gsm
allow=alaw

The call does not come into the context mycontext but into the default
context...

How can I authenticate this call so that it does not go into the default
context ???

Jonas.
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Re: [asterisk-users] Incoming calls coming into default context

[Olle E. Johansson]

21 dec 2009 kl. 12.00 skrev jonas kellens:

 My SIP-provider sends my a SIP-invite like this :
 
 INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0
 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
 Max-Forwards: 70
 From: sip:321445x...@80.xx.xx.69;tag=f395877e02bf8eb2fd8f5a0e
 To: sip:329298y...@80.xx.xx.69
 Call-ID: f395877e02187250fd8f5...@80.xx.xx.68
 CSeq: 1 INVITE
 User-Agent: SysMaster VoIP Gateway v1.2.0
 Contact: sip:321445x...@80.xx.xx.68:5060
 Remote-Party-ID: 
 sip:321445x...@80.xx.xx.69;party=calling;screen=yes;privacy=off
 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE
 Content-Type: application/sdp
 Content-Length: 261
 
 This is my sip.conf :
 
 [outgoing]
 type=peer
 host=sip.XXX.tld
 username=329298xxx6
 secret=my-secret
 fromuser=329298xxx6
 disallow=all
 allow=gsm
 allow=alaw
 
 ; incoming
 [329298yyy6]
 type=user
 host=sip.XXX.tld
 context=mycontext
 disallow=all
 allow=gsm
 allow=alaw
 
 The call does not come into the context mycontext but into the default 
 context...
 
 How can I authenticate this call so that it does not go into the default 
 context ???

Well, you have type=user on the incoming context, so that won't be matched for 
incoming calls. Change it to type=peer and you will be a much happier Asterisk 
user.

/O
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[asterisk-users] Incoming Calls via SIP Trunks

[Andreas M.]

Hello,
i have one question regarding incoming SIP INVITES.

I have a testbed where i have 5 extnsions : 6001 - 6005
Domain : domainA.com

Then i have configured a sip trunk, where my PBX registers to a foreign SIP 
Proxy.
All is working fine, until following scenario:

Incoming call from [EMAIL PROTECTED] (SRV exists,user also exists in pbx as 
extension, but different
domain!)

When i try this, the pbx answers with an proxy-auth.

When i remove extension 6002, all is working again as aspected.

Question: does asterisk not verify the domainpart ? How are incoming INVITES 
processed ?!

I´m using the latest SVN Branch 1.4

br,
Andreas M.

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Re: [asterisk-users] Incoming calls on zaptel not answered.

[Matt Watson]

On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote:
 After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
 zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
 working.

THis isn;t going to fix your problem... but just FYI, you don't need to 
install libpri if you are just using a TDM400P (since its not a PRI / BRI 
[1.6 libpri does BRI as well] card). 

Might save you a little bit of time in the future, and its one less thing to 
consider as a problem.

-- 
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http://www.mattgwatson.ca

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Re: [asterisk-users] Incoming calls on zaptel not answered.

[Jose Flores Galicia]

Hi,
I need libpri, because I have a TE110P E1 with a PRI ISDN service.

2008/7/15 Matt Watson [EMAIL PROTECTED]:

 On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote:
  After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading
 libpri,
  zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
  working.

 THis isn;t going to fix your problem... but just FYI, you don't need to
 install libpri if you are just using a TDM400P (since its not a PRI / BRI
 [1.6 libpri does BRI as well] card).

 Might save you a little bit of time in the future, and its one less thing
 to
 consider as a problem.

 --
 Matt Watson
 http://www.mattgwatson.ca

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-- 
Jose Flores Galicia
[EMAIL PROTECTED]
BriefCode  Code Based Training
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Re: [asterisk-users] Incoming calls on zaptel not answered.

[Jose Flores Galicia]

Thank you,

yes, I changed the PCI Slot and it's the same,
I get a used card from a customer with 2 FXO, same REV, that board was
working on the customer server, put it on mine, and stop working.
I put my board on his server and the board is working perfectly.

I had not  test outgoing calls on that board, I tried and outgoing works
fine.



2008/7/15 Noah Miller [EMAIL PROTECTED]:

 Hi Jose -

  After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading
 libpri,
  zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
  working.
 
  The board is working, I tested in another server with the 1.2.13 asterisk
  version.
  Also changed the pci slot where the board is.

 Hmm.  Bad or incompatible PCI slot?  Can you (at least for testing
 purposes) switch back to the original PCI slot you were using when the
 card worked?


 - Noah

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-- 
Jose Flores Galicia
[EMAIL PROTECTED]
BriefCode  Code Based Training
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[asterisk-users] Incoming calls on zaptel not answered.

[Jose Flores Galicia]

After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
working.

The board is working, I tested in another server with the 1.2.13 asterisk
version.

When a call is incoming, I do a ztmonitor to check the rx and tx values, but
nothing appears on screen.
Also changed the pci slot where the board is.

The zaptel drivers are load properly, I also have a TE110P on the same
server as pri_cpe and has no issues.

Has anyone of you suffered the same? how did you solve it?

Thank you.

-- 
Jose Flores Galicia
[EMAIL PROTECTED]
BriefCode  Code Based Training
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Re: [asterisk-users] Incoming calls on zaptel not answered.

[Noah Miller]

Hi Jose -

 After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
 zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
 working.

 The board is working, I tested in another server with the 1.2.13 asterisk
 version.
 Also changed the pci slot where the board is.

Hmm.  Bad or incompatible PCI slot?  Can you (at least for testing
purposes) switch back to the original PCI slot you were using when the
card worked?


- Noah

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Re: [asterisk-users] incoming calls through callcentric sip account!!

[Emmanuel Favre-Nicolin]

Le vendredi 20 juin 2008, RoLaNd RoLaNd a écrit :
 Hi all,

 i've recently acquired a callcentric account.

 i've perfectly setup my sip.conf and extensions.conf to make outgoing
 calls.

Well, I had the same problem and had to debug. In fact for some reason, and 
it's a bit hackward, incoming call fromm callcentric does not enter in the 
standart extension but in the one corresponding to your username.

I think you have to modify this

[from-callcentric]
exten = s,1,Dial(SIP/107)

to

[from-callcentric]
exten = s,username,Dial(SIP/107)

You can find this using debbuging. 

The problem is that in their support, they told us something wrong, that is to 
use the standart extension, and we trully want to believe them! Why would 
they give us a wrong information?

If it solves your problem, then it should maybe be changed in their support 
section.

What version of asterisk do you use?

I use Asterisk 1.4.20.1. 

If you use asterisk 1.4.x, it could be use a problem with asterisk 1.4 and not 
1.2.x?

By the way, I had problems with asterisk 1.6*, it would simply not work, I 
don't remember why. Did you try asterisk 1.6.x with callcentric?

 but the problem is with incoming calls!  when i call in, asterisk doesnt
 even see the incoming call! how is tht possible!

 please see the following my config:

 sip.conf:


   [general]

   dtmfmode = rfc2833

   context=from-callcentric

   srvlookup=yes

   register = 
 username:[EMAIL PROTECTED]/username

   [callcentric]

   type=peer

   context=from-callcentric

   host=callcentric.com

   username=username

   secret=password

   fromuser=username

   
 fromdomain=callcentric.com

   insecure=very







   [107]

   context=to-callcentric

   type=friend

   username=107

   secret=secret

   host=dynamic


 etensions.conf:



   [from-callcentric]

   exten = 
 s,1,Dial(SIP/107)





   [to-callcentric]

   exten = 
 _0.,1,Dial(SIP/[EMAIL PROTECTED])



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[asterisk-users] incoming calls through callcentric sip account!!

[RoLaNd RoLaNd]

Hi all,

i've recently acquired a callcentric account.

i've perfectly setup my sip.conf and extensions.conf to make outgoing calls.

but the problem is with incoming calls!  when i call in, asterisk doesnt even 
see the incoming call! 
how is tht possible!

please see the following my config:

sip.conf:


[general]

dtmfmode = rfc2833

context=from-callcentric

srvlookup=yes

register = 
username:[EMAIL PROTECTED]/username

[callcentric]

type=peer

context=from-callcentric

host=callcentric.com

username=username

secret=password

fromuser=username


fromdomain=callcentric.com

insecure=very







[107]

context=to-callcentric

type=friend

username=107

secret=secret

host=dynamic


etensions.conf:



[from-callcentric]

exten = 
s,1,Dial(SIP/107)





[to-callcentric]

exten = 
_0.,1,Dial(SIP/[EMAIL PROTECTED])



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Re: [asterisk-users] Incoming calls not being answered by asterisk

[RoLaNd RoLaNd]

Hey thanks for the help :)
though i already did that, and the sip debugging info shows me tht its ringing 
on the respective sip extension (1002) but nothing is happening..
so i guess its true it IS a diala plan issue tht i am yet to figure it out ...



 Date: Sat, 24 May 2008 14:20:45 +0100
 From: [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Incoming calls not being answered by asterisk
 
 The first thing to do is type sip debug on the console and place the
 call from the Sipura. If you get a bunch of SIP messages flashing down
 your console you know the call is reaching Asterisk and it's most
 likely going to be an issue authenticating the call or a problem in
 your dial plan.
 
 If no SIP messages flash up then the call is not reaching your Asterisk 
 server.
 
 Regards,
 
 Greyman.
 
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[asterisk-users] Incoming calls not being answered by asterisk

[RoLaNd RoLaNd]

Hello all,

ive got the following setup currently:

 
   __Sipura 3102-PSTN
  |
Lan | 
  |
  |__asterisk

i configured both asterisk and pstn to be able to receive/make calls through 
each other using sip of course..
but the thing is i want asterisk that when it receives an incoming call from 
sipura, to answer it, play msg that i recorded and wait for the caller to dial 
in an extension, where it would transfer the caller to that exntension, and in 
case the extension owner isnt available to answer it would direct him to his 
voicemail(tht i dont know how to set yet), and in case the caller didnt dial 
any extension in a certain amount of time, it automaticly directs it to a 
specific extensions i'd specify..

i tried the examples given in lots of forums and so on none of them worked, the 
phone keeps on ringing with every incomign dial plan ive specified without 
asterisk answering it..
the thing i did is that sipura directs incoming calls to 1002, so ive set the 
context of 1002 in sip.conf to a dial plan of [incoming-sipura] and ive set the 
commands i mentioned earlier tht i took out of those forums.. but theyre not 
working!!!

anyone has an example i could go on with ? 
any help would be apreciated:)

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Re: [asterisk-users] Incoming calls not being answered by asterisk

[Grey Man]

The first thing to do is type sip debug on the console and place the
call from the Sipura. If you get a bunch of SIP messages flashing down
your console you know the call is reaching Asterisk and it's most
likely going to be an issue authenticating the call or a problem in
your dial plan.

If no SIP messages flash up then the call is not reaching your Asterisk server.

Regards,

Greyman.

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Re: [asterisk-users] Incoming calls not being answered by asterisk

[Roberto Milani]

Ciao Roand

I think you should buy a book and do some reading to build up your  
knowledge.


but in the meantime try something like this in the dialplan  
(extensions.conf)


exten = PSTN,1,Answer() ; Answer inbound calls or internal miss-dials
exten = PSTN,2,Playback(silence/1)
exten = PSTN,3,Background(enter-ext-of-person) ; input an extension
exten = PSTN,n,WaitExten(20) ; Adjust wait, default 5 sec
exten = PSTN,n,Goto(internal,${EXTEN},1) ; Goto the correct extension
exten = PSTN,n,Hangup() ; End the call

where PSTN is your sipura SIP name (1002 i think)

Ciao
Roberto


On May 24, 2008, at 3:09 AM, RoLaNd RoLaNd wrote:


Hello all,

ive got the following setup currently:


   __Sipura 3102-PSTN
  |
Lan |
  |
  |__asterisk

i configured both asterisk and pstn to be able to receive/make calls  
through each other using sip of course..
but the thing is i want asterisk that when it receives an incoming  
call from sipura, to answer it, play msg that i recorded and wait  
for the caller to dial in an extension, where it would transfer the  
caller to that exntension, and in case the extension owner isnt  
available to answer it would direct him to his voicemail(tht i dont  
know how to set yet), and in case the caller didnt dial any  
extension in a certain amount of time, it automaticly directs it to  
a specific extensions i'd specify..


i tried the examples given in lots of forums and so on none of them  
worked, the phone keeps on ringing with every incomign dial plan ive  
specified without asterisk answering it..
the thing i did is that sipura directs incoming calls to 1002, so  
ive set the context of 1002 in sip.conf to a dial plan of [incoming- 
sipura] and ive set the commands i mentioned earlier tht i took out  
of those forums.. but theyre not working!!!


anyone has an example i could go on with ?
any help would be apreciated:)

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[asterisk-users] Incoming calls on PSTN trunk not disconnected (bsnl, india)

[Prashant Sharma]

I am trying to configure Asterisk for BSNL, india network.
I have successfully configured it for outgoing calls.

When any outside number make any call to trunk then it receives the call
properly but when the call is disconnected by inside extension then outside
phone does not get a busy tone.

Asterisk incoming call log:

-- Executing [EMAIL PROTECTED]:2] Dial(Zap/4-1, Zap/1) in new stack
-- Called 1
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
[Jan 17 11:53:54] WARNING[5030]: chan_zap.c:4153 zt_handle_event: Didn't
finish Caller-ID spill.  Cancelling.
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/4-1
-- Native bridging Zap/4-1 and Zap/1-1
-- Hungup 'Zap/1-1'
  == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'


My system information is as follows:

OS and components:

CentOS 4.5
Asterisk 1.4.17
Zaptel 1.4.7.1
Libpri 1.4.3

extensions.conf
[globals]
OUTBOUNDTRUNK=Zap/4

[incoming]
; incoming calls from FXO
exten = s,1,Dial(Zap/1)

[outbound-dialing]
;Outbound dialing
exten = _X.,1,Verbose(1|Outside number|${EXTEN})
exten = _X.,n,Dial(${OUTBOUNDTRUNK}/${EXTEN})

[phones]
include = outbound-dialing

zaptel.conf file:

fxsks=4
fxoks=1
loadzone=in
defaultzone=in

# /sbin/ztcfg -vv this linux command gives following output:

Zaptel Version: 1.4.7.1
Echo Canceller: MG2
Configuration
==

Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

2 channels to configure.

zapata.conf file looks like:
[trunkgroups]
; Define

[channels]
;hardware channels
;default
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
callwaiting=no
immediate=no
cidstart=ring
cidsignalling=dtmf

;define channels
signalling=fxo_ks ;Use FXO signaling for FXS channel
context=phones
channel = 1

signalling=fxs_ks ;Use FXS signaling for FXO channel
context=incoming
channel = 4
Any sort of help will be appreciated.

Thanks in advance

Prashant
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[asterisk-users] Incoming Calls

[Paulo Pinheiro]

I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me an
error message stating the phone number I dialed is not in service. When
I leave the DID number and CLID number blanks it works fine. I really
need to have the system identifying multiple phone numbers ( multiple
trunks ) but I have not been able to do so. Would anyone be able to
help?

 

Thanks,

 

Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767

 

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Re: [asterisk-users] Incoming Calls

[Doug Lytle]

Paulo Pinheiro wrote:

 I am having a problem that I would like to verify if someone could 
 help…I am using bandwith.com as my SIP TRUNK provider. When I place 
 the phone number in the DID number field ( using Elastix) it gives me 
 an error message stating the phone number I dialed is not in service. 
 When I leave the DID number and CLID number blanks



Your best bet would be to ask them. They do have both mailing lists and 
forums.

http://sourceforge.net/mail/?group_id=161807
http://sourceforge.net/forum/?group_id=161807

Doug

-- 
 
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Safety, deserve neither Liberty nor Safety.



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Re: [asterisk-users] Incoming Calls

[Jose P. Espinal]

Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see 
how can we help you?


Regards,



Paulo Pinheiro wrote:


I am having a problem that I would like to verify if someone could 
help...I am using bandwith.com as my SIP TRUNK provider. When I place 
the phone number in the DID number field ( using Elastix) it gives me 
an error message stating the phone number I dialed is not in service. 
When I leave the DID number and CLID number blanks it works fine. I 
really need to have the system identifying multiple phone numbers ( 
multiple trunks ) but I have not been able to do so. Would anyone be 
able to help?


 


Thanks,

 


Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767

 




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Re: [asterisk-users] Incoming Calls

[Zaheer K. Master]

Hi Paulo,

Make sure your DID number is in the e.164 format, ie, +15551234567. I had
the same issue with bandwidth.com and that fixed the problem.

 

HTH,

Zaheer

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 11:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming Calls

 

I am having a problem that I would like to verify if someone could help.I am
using bandwith.com as my SIP TRUNK provider. When I place the phone number
in the DID number field ( using Elastix) it gives me an error message
stating the phone number I dialed is not in service. When I leave the DID
number and CLID number blanks it works fine. I really need to have the
system identifying multiple phone numbers ( multiple trunks ) but I have not
been able to do so. Would anyone be able to help?

 

Thanks,

 

Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767

 

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Re: [asterisk-users] Incoming Calls

[Tony Plack]

 Hi Paulo,

 Make sure your DID number is in the e.164 format, ie, +15551234567.
 I had the same issue with bandwidth.com and that fixed the problem.


 HTH,

 Zaheer

Zaheer is right.  Everything from bandwidth is 164 format.  So you need the 
+15551234567 in the dial plan as well as in your sip.conf

Tony Plack

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Re: [asterisk-users] Incoming Calls

[Paulo Pinheiro]

Hi Jose, I apologize for the lack of information..I am new to this...Let
me try to be more specific:

 

I've got Asterisk installed on Linux. I am using Elastix as the front
end to make changes in the system. 

 

Under the Trunk set up these are my setting for the Peer Details:

 

allow=ulawalawgsm

auth=plaintext

canreinvite=no

context=from-internal

disallow=all

dtmfmode=inband

fromdomain=xxx.xxx.xxx.xxx (IP address)

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=friend

 

these are my settings for User Details:

 

allow=ulaw

canreinvite=no

context=from-sip-external

dtmfmode=rfc2833

host=xxx.xxx.xxx.xxx (IP addres)

nat=no

port=5060

reinvite=no

type=peer

 

When setting up the income routes if I place the phone number in the DID
Number field, when calling the number I receive a message stating the
phone number is not listed or out of service. When I leave the DID
Number field blank everything works because it does a catch all scenario
but that is not what I am looking for.

 

I have tried to place the phone number with +1 in front of it and still
does not work. Any way to help?

 

Thanks much, 

Paulo

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jose P.
Espinal
Sent: Wednesday, January 02, 2008 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls

 

Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see
how can we help you?

Regards,



Paulo Pinheiro wrote: 

I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me an
error message stating the phone number I dialed is not in service. When
I leave the DID number and CLID number blanks it works fine. I really
need to have the system identifying multiple phone numbers ( multiple
trunks ) but I have not been able to do so. Would anyone be able to
help?

 

Thanks,

 

Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767

 

 






 
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Re: [asterisk-users] Incoming Calls

[Jonn R Taylor]

Paulo,



I am using them also. All call traffic to and from them must be in e164 format. 
So your calls have to look like this, +15615551212 or +011 for international. 
They do not let you set the caller name, but will let you set calling number 
and that also needs to be e164 format.



Jonn



  _

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 10:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming Calls



I am having a problem that I would like to verify if someone could help...I am 
using bandwith.com as my SIP TRUNK provider. When I place the phone number in 
the DID number field ( using Elastix) it gives me an error message stating the 
phone number I dialed is not in service. When I leave the DID number and CLID 
number blanks it works fine. I really need to have the system identifying 
multiple phone numbers ( multiple trunks ) but I have not been able to do so. 
Would anyone be able to help?



Thanks,



Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767





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Re: [asterisk-users] Incoming Calls

[Jonn R Taylor]

allow=ulawalaw

canreinvite=no

context=from-internal

disallow=all

dtmfmode=auto

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=peer



This should work for you. They only accept g711 and g729. There service only 
works with static ip's, so there is no auth used.



Jonn

  _

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



Hi Jose, I apologize for the lack of information..I am new to this...Let me try 
to be more specific:



I've got Asterisk installed on Linux. I am using Elastix as the front end to 
make changes in the system.



Under the Trunk set up these are my setting for the Peer Details:



allow=ulawalawgsm

auth=plaintext

canreinvite=no

context=from-internal

disallow=all

dtmfmode=inband

fromdomain=xxx.xxx.xxx.xxx (IP address)

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=friend



these are my settings for User Details:



allow=ulaw

canreinvite=no

context=from-sip-external

dtmfmode=rfc2833

host=xxx.xxx.xxx.xxx (IP addres)

nat=no

port=5060

reinvite=no

type=peer



When setting up the income routes if I place the phone number in the DID Number 
field, when calling the number I receive a message stating the phone number is 
not listed or out of service. When I leave the DID Number field blank 
everything works because it does a catch all scenario but that is not what I am 
looking for.



I have tried to place the phone number with +1 in front of it and still does 
not work. Any way to help?



Thanks much,

Paulo





From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose P. Espinal
Sent: Wednesday, January 02, 2008 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see how can 
we help you?

Regards,



Paulo Pinheiro wrote:

I am having a problem that I would like to verify if someone could help...I am 
using bandwith.com as my SIP TRUNK provider. When I place the phone number in 
the DID number field ( using Elastix) it gives me an error message stating the 
phone number I dialed is not in service. When I leave the DID number and CLID 
number blanks it works fine. I really need to have the system identifying 
multiple phone numbers ( multiple trunks ) but I have not been able to do so. 
Would anyone be able to help?



Thanks,



Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767











  _






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Re: [asterisk-users] Incoming Calls

[Paulo Pinheiro]

Hi John, I have copied your changes in the Peer Details section of the
trunk set up...then I went ahead and added the DID number in the Income
Routes but still did not work. I tried the number alone and also tried
adding the + sign in front of it. Do you think we should have any
changes in the User Details section of the trunk set up?

 

Thanks much, 

Paulo

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonn R
Taylor
Sent: Wednesday, January 02, 2008 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls

 

allow=ulawalaw

canreinvite=no

context=from-internal

disallow=all

dtmfmode=auto

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=peer

 

This should work for you. They only accept g711 and g729. There service
only works with static ip's, so there is no auth used.

 

Jonn



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo
Pinheiro
Sent: Wednesday, January 02, 2008 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls

 

Hi Jose, I apologize for the lack of information..I am new to this...Let
me try to be more specific:

 

I've got Asterisk installed on Linux. I am using Elastix as the front
end to make changes in the system. 

 

Under the Trunk set up these are my setting for the Peer Details:

 

allow=ulawalawgsm

auth=plaintext

canreinvite=no

context=from-internal

disallow=all

dtmfmode=inband

fromdomain=xxx.xxx.xxx.xxx (IP address)

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=friend

 

these are my settings for User Details:

 

allow=ulaw

canreinvite=no

context=from-sip-external

dtmfmode=rfc2833

host=xxx.xxx.xxx.xxx (IP addres)

nat=no

port=5060

reinvite=no

type=peer

 

When setting up the income routes if I place the phone number in the DID
Number field, when calling the number I receive a message stating the
phone number is not listed or out of service. When I leave the DID
Number field blank everything works because it does a catch all scenario
but that is not what I am looking for.

 

I have tried to place the phone number with +1 in front of it and still
does not work. Any way to help?

 

Thanks much, 

Paulo

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jose P.
Espinal
Sent: Wednesday, January 02, 2008 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls

 

Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see
how can we help you?

Regards,



Paulo Pinheiro wrote: 

I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me an
error message stating the phone number I dialed is not in service. When
I leave the DID number and CLID number blanks it works fine. I really
need to have the system identifying multiple phone numbers ( multiple
trunks ) but I have not been able to do so. Would anyone be able to
help?

 

Thanks,

 

Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767

 

 
 
 
 
 






 
 
 
 
 
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Re: [asterisk-users] Incoming Calls

[Tony Plack]

 Hi John, I have copied your changes in the Peer Details section of
 the trunk set up…then I went ahead and added the DID number in the
 Income Routes but still did not work. I tried the number alone and
 also tried adding the + sign in front of it. Do you think we should
 have any changes in the User Details section of the trunk set up?


 Thanks much,

 Paulo

Paulo,
I think I understand what you are trying to do, but that is not the correct 
location for it.

The DID number is the number calling you.  If you place your trunk number into 
the configuration, you will only get calls on that trunk when the DID is your 
trunk number (not really correct).

What you need, is a dial-plan entry with _+15551234567,n,Noop type of setting 
in the extensions.conf.  What you are changing is the sip.conf file.

I do not know if Elastix supports that type of change directly in the 
extensions.conf, but it should.

Tony Plack

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[asterisk-users] Incoming calls

[Gustavo Gonzalez]

Hello I have a question about incoming calls on TDM400P cards. I want to
know why an incoming call appear in a sorpresive way on a phone that I
pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
channels )that are free. I have four lines connected to my TDM400P card and
when I get a free Zap channel to call I hear the voice of a people on the
other side from an incomming call, I think that asterisk bridge my free
channel with incomming calls but how do this?Thanks for any idea.

Alejandro González
Grupo Gestión
4384-0660
www.grupo-gestion.com.ar
[EMAIL PROTECTED]
---

---
RI 9000-1069
Sistema de Gestión de Calidad
Certificado por IRAM
Norma ISO: 9001-2000
 
 
 
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006
01:45 p.m.
 

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Re: [asterisk-users] Incoming calls

[C F]

Glare that's what it's called, if the number you advertise as your
business number is zap/1 then use zap/G1 to dial out, otherwise use
zap/g1 to dial out. This will reduce but not eliminate the problem.

http://www.telos-systems.com/techtalk/gldefs.htm#Glare


On 10/18/07, Gustavo Gonzalez [EMAIL PROTECTED] wrote:
 Hello I have a question about incoming calls on TDM400P cards. I want to
 know why an incoming call appear in a sorpresive way on a phone that I
 pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
 channels )that are free. I have four lines connected to my TDM400P card and
 when I get a free Zap channel to call I hear the voice of a people on the
 other side from an incomming call, I think that asterisk bridge my free
 channel with incomming calls but how do this?Thanks for any idea.

 Alejandro González
 Grupo Gestión
 4384-0660
 www.grupo-gestion.com.ar
 [EMAIL PROTECTED]
 ---

 ---
 RI 9000-1069
 Sistema de Gestión de Calidad
 Certificado por IRAM
 Norma ISO: 9001-2000



 --
 No virus found in this outgoing message.
 Checked by AVG Free Edition.
 Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006
 01:45 p.m.


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Re: [asterisk-users] incoming calls in SIP

[Dovid B]

It seems that the other end is having an issue authenticating you. I have 
seen lots of switches act up with asterisk if you don't tweak you settings in 
sip.conf just right. Do a SIP debug and have a look at te INVITE request.
  - Original Message - 
  From: Ondrej Polívka 
  To: asterisk-users@lists.digium.com 
  Sent: Saturday, August 18, 2007 3:25 PM
  Subject: [asterisk-users] incoming calls in SIP


  Hi, when I try to call in it tells me: NOTICE[11664]: chan_sip.c:10637 
handle_request_invite: Failed to authenticate user 585415198 sip:[EMAIL 
PROTECTED];tag=as18abefe8 
  Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04. 
Outcoming and internal calls functions well. Thanks 

  sip.conf: 
  [general] 
  callevents=yes 

  register = username:[EMAIL PROTECTED]/username 

  [100] 
  callerid=100 
  secret=password 
  type=friend 
  context=internal 
  srvlookup=yes 
  type=friend 
  qualify=yes 
  nat=no 
  host=dynamic 
  canreinvite=no 
  context=internal 
  call-limit=1 

  [101] 
  callerid=101 
  secret=password 
  type=friend 
  context=internal 
  srvlookup=yes 
  type=friend 
  qualify=yes 
  nat=no 
  host=dynamic 
  canreinvite=no 
  context=internal 
  call-limit=1 

  [TRUNK-587207103] 
  type=friend 
  context=incoming 
  username=username 
  secret=password 
  host=sip.mujtelefon.cz 
  dtmfmode=info 
  canreinvite=no 
  qualify=no 
  nat=yes 
  disallow=all 
  allow=g729 
  allow=gsm 
  allow=alaw 






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[asterisk-users] incoming calls in SIP

[Ondřej Polívka]

 Hi, when I try to call in it tells me: NOTICE[11664]: chan_sip.c:10637
handle_request_invite: Failed to authenticate user 585415198
sip:[EMAIL PROTECTED] sip:[EMAIL PROTECTED];tag=as18abefe8
Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04.
Outcoming and internal calls functions well. Thanks

sip.conf:
[general]
callevents=yes

register = username:[EMAIL PROTECTED]/username

[100]
callerid=100
secret=password
type=friend
context=internal
srvlookup=yes
type=friend
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
call-limit=1

[101]
callerid=101
secret=password
type=friend
context=internal
srvlookup=yes
type=friend
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
call-limit=1

[TRUNK-587207103]
type=friend
context=incoming
username=username
secret=password
host=sip.mujtelefon.cz
dtmfmode=info
canreinvite=no
qualify=no
nat=yes
disallow=all
allow=g729
allow=gsm
allow=alaw
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[asterisk-users] Incoming calls don't arrive for correct number

[Frederico Madeira]

I have an asterisk box registering 100 numbers on a voip provider.
Numers are: 2546.1000 to 2546.1099
My problem is that every incoming call arrived to number 2546.1099 that is
the last number to register on voip provider. The correct is call arrive in
destination number.
See this exaple:
I call to 2546.1000.
-- Executing Dial(SIP/25461099-08738060, Zap/g1/3000) in new stack
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called g1/3000
   -- Zap/1-1 is proceeding passing it to SIP/25461099-08738060
   -- Zap/1-1 is ringing
   -- Zap/1-1 answered SIP/25461099-08738060
   -- Hungup 'Zap/1-1'
 == Spawn extension (default, 25461000, 1) exited non-zero on
'SIP/25461099-08738060'
How i solve this problem ??
See parts of my sip.conf
register=25461000:[EMAIL PROTECTED]/25461000
register= 25461001:[EMAIL PROTECTED]/25461001
register=25461002:[EMAIL PROTECTED]/25461002
register= 25461003:[EMAIL PROTECTED]/25461003
.
.
.
register=25461099:[EMAIL PROTECTED]/25461099
[provider-25461000]
type=friend
context=default
secret=
username=25461000
host=sip.provider.com
fromuser=25461000
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
[provider-25461001]
type=friend
context=default
secret=
username=25461001
host=sip.provider.com
fromuser=25461001
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
[provider-25461002]
type=friend
context=default
secret=
username=25461002
host=sip.provider.com
fromuser=25461002
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
.
.
.
[provider-25461099]
type=friend
context=default
secret=
username=25461099
host=sip.provider.com
fromuser=25461099
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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Re: [asterisk-users] Incoming calls don't arrive for correct number

[Marco Mouta]

your problem is that you need to handle this in your dialpan to achieve
which DID has been dialed! look for SIPGETHEADER application on asterisk,
you shoul look for variable to where it comes the DID

On 11/27/06, Frederico Madeira [EMAIL PROTECTED] wrote:


I have an asterisk box registering 100 numbers on a voip provider.
Numers are: 2546.1000 to 2546.1099
My problem is that every incoming call arrived to number 2546.1099 that is
the last number to register on voip provider. The correct is call arrive in
destination number.
See this exaple:
I call to 2546.1000.
-- Executing Dial(SIP/25461099-08738060, Zap/g1/3000) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/3000
-- Zap/1-1 is proceeding passing it to SIP/25461099-08738060
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/25461099-08738060
-- Hungup 'Zap/1-1'
  == Spawn extension (default, 25461000, 1) exited non-zero on
'SIP/25461099-08738060'
How i solve this problem ??
See parts of my sip.conf
register=25461000:[EMAIL PROTECTED]/25461000
register= 25461001:[EMAIL PROTECTED]/25461001
register=25461002:[EMAIL PROTECTED]/25461002
register= 25461003:[EMAIL PROTECTED]/25461003
.
.
.
register=25461099:[EMAIL PROTECTED]/25461099
[provider-25461000]
type=friend
context=default
secret=
username=25461000
host=sip.provider.com
fromuser=25461000
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
[provider-25461001]
type=friend
context=default
secret=
username=25461001
host=sip.provider.com
fromuser=25461001
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
[provider-25461002]
type=friend
context=default
secret=
username=25461002
host=sip.provider.com
fromuser=25461002
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
.
.
.
[provider-25461099]
type=friend
context=default
secret=
username=25461099
host=sip.provider.com
fromuser=25461099
fromdomain=sip.provider.com
nat=yes
insecure=very
canreinvite=no
qualify=yes
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br

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--
Best regards,

Marco Mouta
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[asterisk-users] Incoming calls, identify

[joea, j4computers]

Just delving into asterisk, using trixbox 1.2 and a TDM400p.  The card will 
have two FXO and two FXS modules.

Two incoming analog lines, which need to be treated as distinct entities.  
Meaning, for example, line 1= company1, line2=company2, or line 1= home line, 
line2=business line.  In my limited setup and testing, did not see (obviously) 
how to do this.  I would think this is fundamental and only my new-ish-ness is 
in the way.

Also, as the card does not have any docs with it, what is the power connector 
for?  Is this necessary to allow ringer power to be supplied to analog phones?

joe

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Re: [asterisk-users] Incoming calls, identify

[Jay R. Ashworth]

On Wed, Sep 20, 2006 at 08:48:17AM -0400, joea, j4computers wrote:
 Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card
 will have two FXO and two FXS modules.

 Two incoming analog lines, which need to be treated as distinct
 entities. Meaning, for example, line 1= company1, line2=company2,
 or line 1= home line, line2=business line. In my limited setup and
 testing, did not see (obviously) how to do this. I would think this is
 fundamental and only my new-ish-ness is in the way.

You'll do that with separate contexts.  If you haven't read The Book

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

do so.  Do so before you ask stupid questions like *I* did, last night.  :-)

 Also, as the card does not have any docs with it, what is the power
 connector for? Is this necessary to allow ringer power to be supplied
 to analog phones?

Correct.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] Incoming calls, identify

[joea, j4computers]

Jay R. Ashworth[EMAIL PROTECTED] Wrote on: 9/20/2006 4:00 PM:
 On Wed, Sep 20, 2006 at 08:48:17AM -0400, joea, j4computers wrote:
 Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card
 will have two FXO and two FXS modules.

 Two incoming analog lines, which need to be treated as distinct
 entities. Meaning, for example, line 1= company1, line2=company2,
 or line 1= home line, line2=business line. In my limited setup and
 testing, did not see (obviously) how to do this. I would think this is
 fundamental and only my new-ish-ness is in the way.
 
 You'll do that with separate contexts.  If you haven't read The Book
 
 http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 
 
 do so.  Do so before you ask stupid questions like *I* did, last night. 
  :-)
 
 Also, as the card does not have any docs with it, what is the power
 connector for? Is this necessary to allow ringer power to be supplied
 to analog phones?
 
 Correct.
 
 Cheers,
 -- jra
 -- 
 Jay R. Ashworth


Thanks for the pointer to what promises to be an excellent resource.

joe

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RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[broadbandvoice]

Try 1.2.3, it works fine.

-- Original message -- From: "James Sturges" [EMAIL PROTECTED]  I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of  the site. It is sending CRC errors )to Telsta, drops all calls once a day  for 1 second, calls getting stuck, quite unpleasant!   I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri.   James-Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED] On Behalf Of Paul C  Sent: Wednesday, 1 March 2006 4:15 PM  To: Asterisk Users Mailing List - Non-Commercial Discussion  Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110pPaul C wrote:   I am running Asterisk 
 1.0.9 
and have been running all my calls through a   VSP over a IAX2 trunk however we have recently purchased and connected a   TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can makeoutgoing calls via it fine, however incoming calls are dropped after a   few seconds ( or as soon as a command like Playback, or the call is   picked up if forwarded to a SIP extensions ).SNIP   overlapdial should usually be no in my experience.Okay I've turned that to no with no change. I've just got off the phone to  Optus and apparently they had a client in melbourne last week and they fixed   the problem by turning crc checking off at the optus end. I don't suppose  that was anybody on here ?   ___ &
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[Asterisk-Users] Incoming calls

[Josh]

Hi,

I run an asterisk server. The configuration is very basic.
Here is my question :
When someone calls my phone line, which is connected to an FXO card,
asterisk is answering using the context :

; Incoming calls goes to this default context  :
[incoming-rtc]
include = postes-sip
;
exten = s,1,Goto(menu,1)
exten = s,2,Hangup
;
exten = menu,1,SetVar(count=0)
exten = menu,2,Answer
exten = menu,3,Background(silence/1)
exten = menu,4,Background(josh/welcome-msg)
exten = menu,5,Background(silence/5)
exten = menu,6,SetVar(count=$[${count} + 1])
exten = menu,7,GotoIf($[${count}  1]?4) ; Repeat 3 times
exten = menu,8,Goto(s,2)

When a friend calls, I would like for him to enter a 4 digit password
in order to access to a sub-menu, if  no password is entered, then the
welcome msg is said ...

Any hints on how to do that ??

Thanks a lot !
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Re: [Asterisk-Users] Incoming calls

[Time Bandit]

 When a friend calls, I would like for him to enter a 4 digit password
 in order to access to a sub-menu, if  no password is entered, then the
 welcome msg is said ...

 Any hints on how to do that ??
In your incoming-rtc context, define an extension (let's say 1234)
exten = 1234,1,Authenticate(5894)
exten = 1234,2,Goto(submenu-context,s,1)

see http://www.voip-info.org/wiki-Asterisk+cmd+authenticate

hth
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Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[pdhales]

Just trying to think - are you using the standard E1 setup from ATP?

I have found that the settings on their website work pretty well.

Also - have you tried to put an answer in your dialplan? That might keep the
dialplan open..

later,

PaulH

- Original Message - 
From: Paul C [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 01, 2006 5:15 PM
Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p


  Paul C wrote:
  I am running Asterisk 1.0.9 and have been running all my calls through
a
  VSP over a IAX2 trunk however we have recently purchased and connected
a
  TE110p to a PRI ( E1 with 16 voice channels ) through Optus.   I can
make
  outgoing calls via it fine, however incoming calls are dropped after a
  few seconds ( or as soon as a command like Playback, or the call is
  picked up if forwarded to a SIP extensions ).

  SNIP 

 
  overlapdial should usually be no in my experience.


 Okay I've turned that to no with no change.  I've just got off the phone
to
 Optus and apparently they had a client in melbourne last week and they
fixed
 the problem by turning crc checking off at the optus end.  I don't suppose
 that was anybody on here ?

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Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[Paul C]

Funnily enough upgrading to 1.2.x solved my problems!  Well that and optus 
changing some stuff as well. zaptel-trunk drivers also helped a lot with 
my echo problems.


- Original Message - 
From: James Sturges [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Sunday, March 05, 2006 6:52 AM
Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p



I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of
the site.  It is sending CRC errors )to Telsta, drops all calls once a day
for 1 second, calls getting stuck, quite unpleasant!

I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri.

James


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul C
Sent: Wednesday, 1 March 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p


Paul C wrote:

I am running Asterisk 1.0.9 and have been running all my calls through a
VSP over a IAX2 trunk however we have recently purchased and connected a
TE110p to a PRI ( E1 with 16 voice channels ) through Optus.   I can 
make



outgoing calls via it fine, however incoming calls are dropped after a
few seconds ( or as soon as a command like Playback, or the call is
picked up if forwarded to a SIP extensions ).



SNIP 




overlapdial should usually be no in my experience.



Okay I've turned that to no with no change.  I've just got off the phone 
to
Optus and apparently they had a client in melbourne last week and they 
fixed


the problem by turning crc checking off at the optus end.  I don't suppose
that was anybody on here ?

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RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[James Sturges]

I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of
the site.  It is sending CRC errors )to Telsta, drops all calls once a day
for 1 second, calls getting stuck, quite unpleasant!

I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri.

James


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul C
Sent: Wednesday, 1 March 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

 Paul C wrote:
 I am running Asterisk 1.0.9 and have been running all my calls through a 
 VSP over a IAX2 trunk however we have recently purchased and connected a 
 TE110p to a PRI ( E1 with 16 voice channels ) through Optus.   I can make

 outgoing calls via it fine, however incoming calls are dropped after a 
 few seconds ( or as soon as a command like Playback, or the call is 
 picked up if forwarded to a SIP extensions ).

 SNIP 


 overlapdial should usually be no in my experience.


Okay I've turned that to no with no change.  I've just got off the phone to 
Optus and apparently they had a client in melbourne last week and they fixed

the problem by turning crc checking off at the optus end.  I don't suppose 
that was anybody on here ? 

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Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[pdhales]

Funnily - I have set up 2 or 3 pri's over the last few weeks on 1.2x and
haven't had any issues.
(and one of those is a high load situation - passthru at an outbound call
centre)

PaulH
Melbourne

- Original Message - 
From: James Sturges [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Sunday, March 05, 2006 7:52 AM
Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p


 I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of
 the site.  It is sending CRC errors )to Telsta, drops all calls once a day
 for 1 second, calls getting stuck, quite unpleasant!

 I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri.

 James


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Paul C
 Sent: Wednesday, 1 March 2006 4:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

  Paul C wrote:
  I am running Asterisk 1.0.9 and have been running all my calls through
a
  VSP over a IAX2 trunk however we have recently purchased and connected
a
  TE110p to a PRI ( E1 with 16 voice channels ) through Optus.   I can
make

  outgoing calls via it fine, however incoming calls are dropped after a
  few seconds ( or as soon as a command like Playback, or the call is
  picked up if forwarded to a SIP extensions ).

  SNIP 

 
  overlapdial should usually be no in my experience.


 Okay I've turned that to no with no change.  I've just got off the phone
to
 Optus and apparently they had a client in melbourne last week and they
fixed

 the problem by turning crc checking off at the optus end.  I don't suppose
 that was anybody on here ?

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[Asterisk-Users] incoming calls dropout on PRI over TE110p

[Paul C]

I am running Asterisk 1.0.9 and have been running 
all my calls througha VSP over a IAX2 trunk however we have recently 
purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through 
Optus. I can make outgoing calls via it fine, however incoming calls 
are dropped after a few seconds ( or as soon as a command like Playback, or the 
call is picked up if forwarded to a SIP extensions ). 

I have upgraded libpri and zaptel to trunk, but I 
don't want to upgrade Asterisk to 1.2 until I've got this all sorted, one 
problem at a time!

Here are my configs :

/etc/zaptel.conf

# Global data

span=1,1,0,ccs,hdb3,crc4bchan=1-15,17dchan=16unused=18-31
loadzone = audefaultzone = au

/etc/asterisk/zapata.conf

[channels]context=from-onramp

overlapdial=yespriindication = 
outofbandswitchtype=euroisdnpridialplan=unknownprilocaldialplan=unknownsignalling=pri_cpe

echocancelwhenbridged=yesechocancel=128echotraining=800

rxgain=5 ; 0txgain=-4.5 ; 
0busydetect=nopridialplan=localinternationalprefix=0011nationalprefix=0usecallerid=yeshidecallerid=nocallprogress=no

group=0channel = 1-15,17

/etc/asterisk/extensions:

[from-onramp];exten = 
s,1,Playback(custom/aa_1)exten = s,1,Dial(SIP/116)exten = 
h,1,Hangup
and here's some log info:

asterisk*CLI pri intense debug span 
1Enabled EXTENSIVE debugging on span 
1asterisk*CLIasterisk*CLIasterisk*CLI 
-- Going to extension s|1 because of Complete received -- 
Executing Dial("Zap/1-1", "SIP/116") in new stack -- 
Called 116 -- Accepting call from '' to 's' on channel 
0/1, span 1 -- SIP/116-5a95 is 
ringing 
-- SIP/116-5a95 answered Zap/1-1 == Spawn extension (from-onramp, s, 
1) exited non-zero on 'Zap/1-1' -- Executing 
Hangup("Zap/1-1", "") in new stack == Spawn extension (from-onramp, h, 
1) exited non-zero on 'Zap/1-1' -- Hungup 
'Zap/1-1'
and going straight to a Playback command rather 
than SIP extension:

asterisk*CLI pri intense debug span 
1Enabled EXTENSIVE debugging on span 
1%%% 
-- Going to extension s|1 because of Complete received -- 
Executing Answer("Zap/2-1", "") in new stack -- Accepting 
call from '' to 's' on channel 0/2, span 1 == Spawn extension 
(from-onramp, s, 1) exited non-zero on 'Zap/2-1' -- 
Executing Hangup("Zap/2-1", "") in new stack == Spawn extension 
(from-onramp, h, 1) exited non-zero on 'Zap/2-1' -- Hungup 
'Zap/2-1'


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Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[Eric \ManxPower\ Wieling]

Paul C wrote:
I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus.   I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ).   


I have upgraded libpri and zaptel to trunk, but I don't want to upgrade 
Asterisk to 1.2 until I've got this all sorted,  one problem at a time!

Here are my configs :

/etc/zaptel.conf

# Global data

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17
dchan=16
unused=18-31
loadzone = au
defaultzone = au

/etc/asterisk/zapata.conf

[channels]
context=from-onramp

overlapdial=yes


overlapdial should usually be no in my experience.
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Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p

[Paul C]

Paul C wrote:
I am running Asterisk 1.0.9 and have been running all my calls through a 
VSP over a IAX2 trunk however we have recently purchased and connected a 
TE110p to a PRI ( E1 with 16 voice channels ) through Optus.   I can make 
outgoing calls via it fine, however incoming calls are dropped after a 
few seconds ( or as soon as a command like Playback, or the call is 
picked up if forwarded to a SIP extensions ).



SNIP 




overlapdial should usually be no in my experience.



Okay I've turned that to no with no change.  I've just got off the phone to 
Optus and apparently they had a client in melbourne last week and they fixed 
the problem by turning crc checking off at the optus end.  I don't suppose 
that was anybody on here ? 


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[Asterisk-Users] Incoming Calls Getting Crossed - Weird

[Michael J. Liberatore]

Hey, I got a weird 
one for you guys,I am running vanilla 1.2.4 and have all incoming 
calls come in as SIP from teliax. Twice over the past week 2 callers who 
have called in around the same time end up talking to each other instead of 
going through the ivr or at some point during the IVR. One said, yeah i 
was talking to another patient and we had a convo. I have double checked 
the dialplan and the logs and everything looks ok. Is this a possible bug 
or can someone tell me what i might be missing? Its very odd but luckily 
fairly rare so far, i am worried it could get worse though.

-Mike
Mike240se



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RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

[Alexander Lopez]

You have stumbled across the new undocumented feature 
app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you 
have any SNOM phones,

  
  
  Hey, I got a weird one 
  for you guys,I am running vanilla 1.2.4 and have all incoming 
  calls come in as SIP from teliax. Twice over the past week 2 callers who 
  have called in around the same time end up talking to each other instead of 
  going through the ivr or at some point during the IVR. One said, yeah i 
  was talking to another patient and we had a convo. I have double checked 
  the dialplan and the logs and everything looks ok. Is this a possible 
  bug or can someone tell me what i might be missing? Its very odd but 
  luckily fairly rare so far, i am worried it could get worse 
  though.
  
  -Mike
  Mike240se
  
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RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

[Michael J. Liberatore]

LMAO! 
app_PatientDatingService

Yes I have all Snom 360's, are you thinking the problem 
isnt asterisk but instead a problem with the Snom phone?
I am running 5.3 on 3 and 5.3.3 on another, i could try 
5.3.6 if you think its the snoms causing the problem...

-mike




From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
LopezSent: Monday, February 20, 2006 6:14 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[Asterisk-Users] Incoming Calls Getting Crossed - Weird

You have stumbled across the new undocumented feature 
app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you 
have any SNOM phones,

  
  
  Hey, I got a weird one 
  for you guys,I am running vanilla 1.2.4 and have all incoming 
  calls come in as SIP from teliax. Twice over the past week 2 callers who 
  have called in around the same time end up talking to each other instead of 
  going through the ivr or at some point during the IVR. One said, yeah i 
  was talking to another patient and we had a convo. I have double checked 
  the dialplan and the logs and everything looks ok. Is this a possible 
  bug or can someone tell me what i might be missing? Its very odd but 
  luckily fairly rare so far, i am worried it could get worse 
  though.
  
  -Mike
  Mike240se
  
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RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

[Alexander Lopez]

It is not the firmware but a setting. "Call Join on 
Xfer (2 calls)"

Make 
sure that is is set to OFF.

SNOMS 
are great ophone but 'features' like this drive me crazy.

Alex



  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Michael 
  J. LiberatoreSent: Monday, February 20, 2006 6:38 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] Incoming Calls Getting Crossed - Weird
  
  LMAO! 
  app_PatientDatingService
  
  Yes I have all Snom 360's, are you thinking the problem 
  isnt asterisk but instead a problem with the Snom phone?
  I am running 5.3 on 3 and 5.3.3 on another, i could try 
  5.3.6 if you think its the snoms causing the problem...
  
  -mike
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
  LopezSent: Monday, February 20, 2006 6:14 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] Incoming Calls Getting Crossed - Weird
  
  You have stumbled across the new undocumented feature 
  app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you 
  have any SNOM phones,
  


Hey, I got a weird 
one for you guys,I am running vanilla 1.2.4 and have all 
incoming calls come in as SIP from teliax. Twice over the past week 2 
callers who have called in around the same time end up talking to each other 
instead of going through the ivr or at some point during the IVR. One 
said, yeah i was talking to another patient and we had a convo. I have 
double checked the dialplan and the logs and everything looks ok. Is 
this a possible bug or can someone tell me what i might be missing? 
Its very odd but luckily fairly rare so far, i am worried it could get worse 
though.

-Mike
Mike240se

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RE: [Asterisk-Users] Incoming Calls Getting Crossed - Weird

[Michael J. Liberatore]

Oh yeah that feature is already off They dont do 
transfers much so it probably didnt happen during a 
transfer.

mike



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
LopezSent: Monday, February 20, 2006 6:44 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[Asterisk-Users] Incoming Calls Getting Crossed - Weird

It is not the firmware but a setting. "Call Join on 
Xfer (2 calls)"

Make 
sure that is is set to OFF.

SNOMS 
are great ophone but 'features' like this drive me crazy.

Alex



  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Michael 
  J. LiberatoreSent: Monday, February 20, 2006 6:38 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] Incoming Calls Getting Crossed - Weird
  
  LMAO! 
  app_PatientDatingService
  
  Yes I have all Snom 360's, are you thinking the problem 
  isnt asterisk but instead a problem with the Snom phone?
  I am running 5.3 on 3 and 5.3.3 on another, i could try 
  5.3.6 if you think its the snoms causing the problem...
  
  -mike
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
  LopezSent: Monday, February 20, 2006 6:14 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] Incoming Calls Getting Crossed - Weird
  
  You have stumbled across the new undocumented feature 
  app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you 
  have any SNOM phones,
  


Hey, I got a weird 
one for you guys,I am running vanilla 1.2.4 and have all 
incoming calls come in as SIP from teliax. Twice over the past week 2 
callers who have called in around the same time end up talking to each other 
instead of going through the ivr or at some point during the IVR. One 
said, yeah i was talking to another patient and we had a convo. I have 
double checked the dialplan and the logs and everything looks ok. Is 
this a possible bug or can someone tell me what i might be missing? 
Its very odd but luckily fairly rare so far, i am worried it could get worse 
though.

-Mike
Mike240se

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[Asterisk-Users] Incoming calls grind to a halt

[David Craigon]

Hi there everybody,
We are running Asterisk 1.2.1 with a TE410P card attached to one
PRI ISDN line, and many SIP phones. Yesterday we ended up in a situation
where all incoming calls were giving the engaged tone. Every time some
tried to ring in we got:

Jan  4 14:56:32 WARNING[896] chan_zap.c: Ring requested on channel 0/5
already in use on span 4.  Hanging up owner.

This happens even though no calls were being made or received. This
happens semi-regularly after use of the server for a while. A reboot
solves the problem.

Dialling out works fine.

I appreciate I haven't given enough information- I've hesitated from
attaching the log file due to it's huge size. I'm not sure which bits
are pertinent.

Can anyone offer any advice?

Thanks,
David
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Re: [Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

[Joerg Lauer]

Hi,

I'm not really sure if this helps you, but as far as I remember, the 
diastring with chan_capi-cm-0.6 is not 
CAPI/g1/0299546476:b${EXTEN},30,r but 
CAPI/ggroup/destination[/params] or in your case 
CAPI/g1/${EXTEN}/b,30,r.


To set your CallerPresentation, use the SetCallerPres() in your 
Dialplan, which is now used as the CLIP/CLIR.


Regards Jörg

Esteban Guana-Jarrin wrote:
Can anyone please provide some help. I have installed an AVM fritz card 
on an asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the 
card driver and chan_capi-cm-0.6. According to the installations guide I 
can now see that the CAPI channel in asterisk is up,


*CLI capi info
Contr1: 2 B channels total, 2 B channels free.

I set up a trunk and the dialstring includes the following,

CAPI/g1/0299546476:b${EXTEN},30,r

My capi.conf is,

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law

; interface sections ...

[ISDN1]  ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, ptp should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * == any
;controller=0;ISDN4BSD default
;controller=7;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1  ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on  ;enable/disable software dtmf detection, recommended 
for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf 
detection

accountcode= ;Asterisk accountcode to use in CDRs
context=from-trunk
;context=capi-in  ;context for incoming calls
holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be 
used. If
;set to 'local' (default value), no hold is done and 
Asterisk may

;play MOH.
;immediate=yes   ;immediate start of pbx with extension 's' if no digits 
were

;received on incoming call (no destination number yet)
;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 
(necessary for older eicon drivers)

;echotail=64 ;echo cancel tail setting
;bridge=yes  ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels 
are busy

devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)

I can't see that a number is assigned to msn, but I read somewhere on 
this list that for this latest version of chan_capi this is not required.


I connected the asterisk box to the ISDN line, which belongs to a Hunt 
group with number as shown in the dialstring and when ringing that 
number from an external line I do not get any tone and asterisk does not 
log any indications of incoming calls via the CAPI channel


Can anyone please shed some light on what do I need to do in order to be 
able to receive calls via this setup.



Thanks in advance,

PolAUs

_
SEEK: Over 80,000 jobs across all industries at Australia's #1 job 
site.http://ninemsn.seek.com.au?hotmail


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Re: [Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

[Armin Schindler]

On Wed, 26 Oct 2005, Esteban Guana-Jarrin wrote:
 Can anyone please provide some help. I have installed an AVM fritz card on an
 asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card 
 driver and
 chan_capi-cm-0.6. According to the installations guide I can now see that the
 CAPI channel in asterisk is up,
 
 *CLI capi info
 Contr1: 2 B channels total, 2 B channels free.
 
 I set up a trunk and the dialstring includes the following,
 
 CAPI/g1/0299546476:b${EXTEN},30,r
 
 My capi.conf is,
 
 [general]
 nationalprefix=0
 internationalprefix=00
 rxgain=0.8
 txgain=0.8
 ;ulaw=yes;set this, if you live in u-law world instead of a-law
 
 ; interface sections ...
 
 [ISDN1]  ;this example interface gets name 'ISDN1' and may be any
 ;name not starting with 'g' or 'contr'.
 ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
 isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
 ;when using NT-mode, ptp should be set in any case
 incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * == any
 ; controller=0;ISDN4BSD default
 ; controller=7;ISDN4BSD USB default
 controller=1 ;capi controller number to use
 group=1  ;dialout group
 ;prefix=0;set a prefix to calling number on incoming calls
 softdtmf=on  ;enable/disable software dtmf detection, recommended for AVM
 cards
 relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
 accountcode= ;Asterisk accountcode to use in CDRs
 context=from-trunk
 ;context=capi-in  ;context for incoming calls
 holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be used.
 If
 ;set to 'local' (default value), no hold is done and Asterisk
 may
 ;play MOH.
 ;immediate=yes   ;immediate start of pbx with extension 's' if no digits were
 ;received on incoming call (no destination number yet)
 ; echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
 ; echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
 ;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
 echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for
 older eicon drivers)
 ; echotail=64 ;echo cancel tail setting
 ; bridge=yes  ;native bridging (CAPI line interconnect) if available
 ; callgroup=1 ;Asterisk call group
 ; deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are 
 busy
 devices=2;number of concurrent calls on this controller
 ;(2 makes sense for single BRI, 30 for PRI)
 
 I can't see that a number is assigned to msn, but I read somewhere on this
 list that for this latest version of chan_capi this is not required.

With 'incomingmsn=*' chan_capi will signal all incoming calls to Asterisk. 
Rules in extensions.conf then decide what to do with the calls.
For outgoing calls, you need to set the correct MSN as callerid.
 
 I connected the asterisk box to the ISDN line, which belongs to a Hunt group
 with number as shown in the dialstring and when ringing that number from an
 external line I do not get any tone and asterisk does not log any indications
 of incoming calls via the CAPI channel
 
 Can anyone please shed some light on what do I need to do in order to be able
 to receive calls via this setup.

What exactly is your problem?

Armin
 
 Thanks in advance,
 
 PolAUs
 
 _
 SEEK: Over 80,000 jobs across all industries at Australia's #1 job site.
 http://ninemsn.seek.com.au?hotmail
 
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Re: [Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

[Esteban Guana-Jarrin]

Armin, thanks for your response

My problem now is that having the configuration on capi.conf as shown in my 
original post I am not able to receive incoming neither make outgoing calls.


When making an outgoing call  I get the following debug and verbose output 
from asterisk,


-- Goto (macro-dialout-trunk,s,24)
   -- Executing Dial(SIP/123-da67, CAPI/g1/0299546476:b,30,r) in new 
stack

   data = g1/0299546476:b,30,r
   capi request group = 2
 == ISDN1: Call CAPI/ISDN1/0299546476:b,30,r-9   (pres=0x00, ton=0x00)
   -- Called g1/0299546476:b,30,r
   -- ISDN1: received CONNECT_CONF PLCI = 0x101
 == ISDN1: Interface cleanup PLCI=0x101
 == No one is available to answer at this time
   -- Executing Goto(SIP/123-da67, s-NOANSWER|1) in new stack
   -- Goto (macro-dialout-trunk,s-NOANSWER,1)

In extensions_additional.conf i've got,

exten = _1.,1,Macro(dialout-trunk,12,${EXTEN:1},)

And in capi.conf i added msn=0299546476.

i don't know why it goes to the a message saying all lines are congested 
now, when the server is connected to the ISDN line (onramp2) with prime 
number as shown in the output. Can you let me know what you understand from 
the output and what you think I'm not doing correct?



For the incoming calls since I have 'incomingmsn=*' then I should get 
chan_capi signalling incoming
calls to Asterisk, I included context=from trunk in capi.conf which should 
route the call to an internal extension, but when I ring the msn 0299546476, 
I can't hear anything except a tone dropping the call.



Armin,  I will appreciate if you can put me in the right direction?

Cheers

PolAus


From: Armin Schindler [EMAIL PROTECTED]
To: Esteban Guana-Jarrin [EMAIL PROTECTED]
CC: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card
Date: Wed, 26 Oct 2005 11:39:05 +0200 (CEST)

On Wed, 26 Oct 2005, Esteban Guana-Jarrin wrote:
 Can anyone please provide some help. I have installed an AVM fritz card 
on an
 asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card 
driver and
 chan_capi-cm-0.6. According to the installations guide I can now see 
that the

 CAPI channel in asterisk is up,

 *CLI capi info
 Contr1: 2 B channels total, 2 B channels free.

 I set up a trunk and the dialstring includes the following,

 CAPI/g1/0299546476:b${EXTEN},30,r

 My capi.conf is,

 [general]
 nationalprefix=0
 internationalprefix=00
 rxgain=0.8
 txgain=0.8
 ;ulaw=yes;set this, if you live in u-law world instead of a-law

 ; interface sections ...

 [ISDN1]  ;this example interface gets name 'ISDN1' and may be 
any

 ;name not starting with 'g' or 'contr'.
 ;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
 isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward 
dial)

 ;when using NT-mode, ptp should be set in any case
 incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * == 
any

 ; controller=0;ISDN4BSD default
 ; controller=7;ISDN4BSD USB default
 controller=1 ;capi controller number to use
 group=1  ;dialout group
 ;prefix=0;set a prefix to calling number on incoming calls
 softdtmf=on  ;enable/disable software dtmf detection, recommended 
for AVM

 cards
 relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf 
detection

 accountcode= ;Asterisk accountcode to use in CDRs
 context=from-trunk
 ;context=capi-in  ;context for incoming calls
 holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be 
used.

 If
 ;set to 'local' (default value), no hold is done and 
Asterisk

 may
 ;play MOH.
 ;immediate=yes   ;immediate start of pbx with extension 's' if no digits 
were

 ;received on incoming call (no destination number yet)
 ; echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
 ; echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
 ;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
 echocancelold=yes;use facility selector 6 instead of correct 8 
(necessary for

 older eicon drivers)
 ; echotail=64 ;echo cancel tail setting
 ; bridge=yes  ;native bridging (CAPI line interconnect) if available
 ; callgroup=1 ;Asterisk call group
 ; deflect=1234567 ;deflect incoming calls to 1234567 if all B channels 
are

 busy
 devices=2;number of concurrent calls on this controller
 ;(2 makes sense for single BRI, 30 for PRI)

 I can't see that a number is assigned to msn, but I read somewhere on 
this

 list that for this latest version of chan_capi this is not required.

With 'incomingmsn=*' chan_capi will signal all incoming calls to Asterisk.
Rules in extensions.conf then decide what to do with the calls.
For outgoing calls, you need to set the correct MSN as callerid.

 I connected the asterisk box to the ISDN line, which belongs to a Hunt 
group
 with number as shown in the dialstring and when ringing that number

[Asterisk-Users] Incoming calls via CAPI and AVM Fritz Card

[Esteban Guana-Jarrin]

Can anyone please provide some help. I have installed an AVM fritz card on 
an asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card 
driver and chan_capi-cm-0.6. According to the installations guide I can now 
see that the CAPI channel in asterisk is up,


*CLI capi info
Contr1: 2 B channels total, 2 B channels free.

I set up a trunk and the dialstring includes the following,

CAPI/g1/0299546476:b${EXTEN},30,r

My capi.conf is,

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law

; interface sections ...

[ISDN1]  ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes  ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, ptp should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * == any
;controller=0;ISDN4BSD default
;controller=7;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1  ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on  ;enable/disable software dtmf detection, recommended for 
AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf 
detection

accountcode= ;Asterisk accountcode to use in CDRs
context=from-trunk
;context=capi-in  ;context for incoming calls
holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be 
used. If
;set to 'local' (default value), no hold is done and 
Asterisk may

;play MOH.
;immediate=yes   ;immediate start of pbx with extension 's' if no digits 
were

;received on incoming call (no destination number yet)
;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary 
for older eicon drivers)

;echotail=64 ;echo cancel tail setting
;bridge=yes  ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are 
busy

devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)

I can't see that a number is assigned to msn, but I read somewhere on this 
list that for this latest version of chan_capi this is not required.


I connected the asterisk box to the ISDN line, which belongs to a Hunt group 
with number as shown in the dialstring and when ringing that number from an 
external line I do not get any tone and asterisk does not log any 
indications of incoming calls via the CAPI channel


Can anyone please shed some light on what do I need to do in order to be 
able to receive calls via this setup.



Thanks in advance,

PolAUs

_
SEEK: Over 80,000 jobs across all industries at Australia's #1 job site.
http://ninemsn.seek.com.au?hotmail


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[Asterisk-Users] Incoming Calls causing Protocol Error (6)

[Douglas Lane]

Hi Everyone,

Got a setup as follows:

Telco  Siemens HiCom 300E  Asterisk1 IAX2 Trunk
Asterisk2  Siemens HiPath 4xxx

The solution works except for one problem. Incoming calls from the telco get
redirected to the Asterisk1 box with the correct extention, only if there is
a callerid set on the call, the Asterisk1 box drops the call (it doesn't
even get to asterisk) with a Unable to handle pre-handled call and
Protocol Error (6). If you disable your callerid on your phone and phone
again via the telco, it gets passed through. Asterisk1 reports Accepting
overlap call from '' to '5804'

Currently using ECMA.1 on the Siemens HiCom 300E, and Asterisk1 is setup
using euroisdn. I am using Asterisk 1.2.0-Beta1. Asterisk1 is running as
pri_cpe as well as secondary sync source.

Any ideas on how to fix this problem?

Would it be better to change the switchtype to Q.SIG on Asterisk and on the
Siemens HiCom 300E ?

Or am I missing a configuration line?

Thanks in advance.
Doug.


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[Asterisk-Users] Incoming calls

[FaberK]

Hi,
stupid question:
how can I let to call an extensions from outside?
Untill now, I've just the possibility to call our number and then,
after the system answer, dial the extension.
My sistem is like this:
SER - internal extensions
Asterisk - incoming/outgoing gateway.

FaberK
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.:FaberK:.
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[Asterisk-Users] incoming calls

[salahssaid2.salah]

hi ;

our * handle good the outgoing calls but 4 incaming calls we have this msg:
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??,

thx in advance,



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[Asterisk-Users] incoming calls

[ali kia]

hi ;



our * handle good the outgoing calls but 4 incaming calls we have this msg:

Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??,



thx in advance,
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Re: [Asterisk-Users] incoming calls

[Andres Tello Abrego]

youa re using -v option multiple times at startup.
That message is perfectly fine.



ali kia wrote:



hi ;

 

our  * handle good  the outgoing calls but 4 incaming calls we have this 
msg :



Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 
(Ring/Answered)...


someone have an idea ??,

 


thx in advance,



http://secure.caramail.lycos.fr/services/content/advdetail.jsp?advid=advprotekonadvsvc=advsecureTARGETCODE=FR_footermail_link 
CaraMail met en oeuvre un nouveau *Concept de Sécurité Globale* 
http://secure.caramail.lycos.fr/services/content/advdetail.jsp?advid=advprotekonadvsvc=advsecureTARGETCODE=FR_footermail_link 
à partir de 1,49 euros par mois





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Re: Re: [Asterisk-Users] incoming calls

[salahssaid2.salah]

 
thk u very match 4 the answer ;
   we have use asterisk -v as u said in startup but we have always 
the same problem
in the begining we used asterisk -vvcv.
   we like know too if the order of v v c v  is important

thanx again


 De: Andres Tello Abrego [EMAIL PROTECTED]
 A: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Objet: Re: [Asterisk-Users] incoming calls
 Date: Fri, 22 Jul 2005 06:53:19 +

 youa re using -v option multiple times at startup.
 That message is perfectly fine.
 
 
 
 ali kia wrote:
  
  
  hi ;
  
   
  
  our  * handle good  the outgoing calls but 4 incaming calls we have this 
  msg :
  
  
  Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 
  (Ring/Answered)...
  
  someone have an idea ??,
  
   
  
  thx in advance,
  
  
  
  a 
  href=http://secure.caramail.lycos.fr/http://secure.caramail.lycos.fr//a/services/content/advdetail.jsp?advid=advprotekonadvsvc=advsecureTARGETCODE=FR_footermail_link
   
  CaraMail met en oeuvre un nouveau *Concept de Sécurité Globale* 
  a 
  href=http://secure.caramail.lycos.fr/http://secure.caramail.lycos.fr//a/services/content/advdetail.jsp?advid=advprotekonadvsvc=advsecureTARGETCODE=FR_footermail_link
   
  à partir de 1,49 euros par mois
  
  
  
  
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Re: Re: [Asterisk-Users] incoming calls

[Tzafrir Cohen]

On Fri, Jul 22, 2005 at 04:41:01PM +, salahssaid2.salah wrote:

  From: Andres Tello Abrego [EMAIL PROTECTED]
  Date: Fri, 22 Jul 2005 06:53:19 +
 
  youa re using -v option multiple times at startup.
  That message is perfectly fine.

And thus see quite a few messages that are not necessarily errors.

 thk u very match 4 the answer ;
 we have use asterisk -v as u said in startup but we have always the 
 same problem in the begining we used asterisk -vvcv.
 we like know too if the order of v v c v  is important
 

No. The order is not important. 

-- 
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http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Incoming calls from BudgetPhone.nl

[Peter Raaijmaker]

(this time with subject)

Hello,

I’m trying to get Asterisk to accept incoming calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
tone.
I tried X-lite, which worked perfect, so my modem (with nat) probably is not
the problem.
I did a sip debug and got the following output.
Because I’m new to Asterisk I can’t get the error why this is not working. 
To me it all looks fine, no warnings or what so ever…
 
The settings in sip.conf and extensions.conf are identical to those of
http://www.voip-info.org/tiki-index.php?page=Talkin2ya
 
Does anyone know what I’m doing wrong
 
Thanks,
Peter.
 
 
---
output of sip debug
---
 
11 headers, 0 lines
Reliably Transmitting (no NAT) to 81.23.228.150:5060:
REGISTER sip:budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
From: sip:[EMAIL PROTECTED];tag=as5dc83db4
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0
 
 
---
server*CLI
-- SIP read from 81.23.228.150:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
From: sip:[EMAIL PROTECTED];tag=as5dc83db4
To:
sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.247a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 REGISTER
WWW-Authenticate: Digest realm=budgetphone.nl,
nonce=42d15009299d7652e8da589cee2723af4b6a96ca
Server: Sip EXpress router (0.8.14-5 (i386/linux))
Content-Length: 0
 
 
--- (9 headers 0 lines)---
Responding to challenge, registration to domain/host name budgetphone.nl
12 headers, 0 lines
Reliably Transmitting (no NAT) to 81.23.228.150:5060:
REGISTER sip:budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
From: sip:[EMAIL PROTECTED];tag=as7e56000d
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username=31717110342, realm=budgetphone.nl,
algorithm=MD5, uri=sip:budgetphone.nl,
nonce=42d15009299d7652e8da589cee2723af4b6a96ca,
response=cd69279e6a2512fd48d267ceea3394da, opaque=
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0
 
 
---
server*CLI
-- SIP read from 81.23.228.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
From: sip:[EMAIL PROTECTED];tag=as7e56000d
To:
sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.98b0
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
Contact: sip:[EMAIL PROTECTED]:5060;q=0.00;expires=120
Server: Sip EXpress router (0.8.14-5 (i386/linux))
Content-Length: 0
 
 
--- (9 headers 0 lines)---
Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound
Registration: Expiry for budgetphone.nl is 120 sec (Scheduling
reregistration in 105000 ms)
Destroying call '[EMAIL PROTECTED]'
server*CLI
-- SIP read from 81.23.228.150:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Max-Forwards: 10
Record-Route: sip:[EMAIL PROTECTED];ftag=as47419911;lr=on
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0
Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa
From: 0031172651375
sip:[EMAIL PROTECTED];tag=as47419911
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 10 Jul 2005 16:37:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 345
 
v=0
o=root 26318 26318 IN IP4 212.203.28.2
s=session
c=IN IP4 81.23.228.139
t=0 0
m=audio 36634 RTP/AVP 3 18 5 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 
--- (15 headers 15 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 81.23.228.150 : 5060 (NAT)
Found peer '31717110342'
Reliably Transmitting (NAT) to 81.23.228.150:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=506
0
Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa
From: 0031172651375
sip:[EMAIL PROTECTED];tag=as47419911
To: sip:[EMAIL PROTECTED];tag=as3f35655f
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=555b996d
Content-Length: 0
 
 
---
Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms
server*CLI
-- SIP read from 81.23.228.150:5060:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0
From: 0031172651375
sip:[EMAIL PROTECTED];tag=as47419911
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as3f35655f
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.8.14-5 

Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

[Michiel van Baak]

On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote:
 (this time with subject)
 
 Hello,
 
 I?m trying to get Asterisk to accept incoming calls from budgetphone.nl.
 When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
 tone.
 I tried X-lite, which worked perfect, so my modem (with nat) probably is not
 the problem.
 I did a sip debug and got the following output.
 Because I?m new to Asterisk I can?t get the error why this is not working. 
 To me it all looks fine, no warnings or what so ever?
 ?
 The settings in sip.conf and extensions.conf are identical to those of
 http://www.voip-info.org/tiki-index.php?page=Talkin2ya
 ?
 Does anyone know what I?m doing wrong
 ?

Can you show us the relevant part in sip.conf and
extensions.conf. It is working fine here (cept for audio
quality and stability of the sip registration, I'm trashing
them soon)
If you post it I can compare it with my setup and maybe that
will show us what's going wrong on your setup
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?
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RE: [Asterisk-Users] Incoming calls from BudgetPhone.nl

[Peter Raaijmaker]

Rene,

I believe you're right, when I disable x-ten's stun server my call isn't
coming through anymore.

But now I don't have a solution but an extra problem I'm afraid!

How to make asterisk run with a stun server? 
Do I have to set one up myself or can I use the x-ten server for example?
Or is there a better way to setup asterisk or my router?

Thanks for your help, hopefully you can help me some more!

Peter Raaijmakers.


-Oorspronkelijk bericht-
Van: Rene Kluwen [mailto:[EMAIL PROTECTED] 
Verzonden: zondag 10 juli 2005 19:28
Aan: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

Long short,

Maybe X-Ten has an stun relay setup and Asterisk doesn't?

Rene Kluwen
Chimit

 (this time with subject)

 Hello,

 I’m trying to get Asterisk to accept incoming calls from budgetphone.nl.
 When I dial my budgetphone nr on a PSTN KPN line it immediately gives a
 busy
 tone.
 I tried X-lite, which worked perfect, so my modem (with nat) probably is
 not
 the problem.
 I did a sip debug and got the following output.
 Because I’m new to Asterisk I can’t get the error why this is not working.
 To me it all looks fine, no warnings or what so ever…
  
 The settings in sip.conf and extensions.conf are identical to those of
 http://www.voip-info.org/tiki-index.php?page=Talkin2ya
  
 Does anyone know what I’m doing wrong
  
 Thanks,
 Peter.
  
  
 ---
 output of sip debug
 ---
  
 11 headers, 0 lines
 Reliably Transmitting (no NAT) to 81.23.228.150:5060:
 REGISTER sip:budgetphone.nl SIP/2.0
 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
 From: sip:[EMAIL PROTECTED];tag=as5dc83db4
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 REGISTER
 User-Agent: Asterisk PBX
 Expires: 120
 Contact: sip:[EMAIL PROTECTED]
 Event: registration
 Content-Length: 0
  
  
 ---
 server*CLI
 -- SIP read from 81.23.228.150:5060:
 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
 From: sip:[EMAIL PROTECTED];tag=as5dc83db4
 To:
 sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.247a
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 REGISTER
 WWW-Authenticate: Digest realm=budgetphone.nl,
 nonce=42d15009299d7652e8da589cee2723af4b6a96ca
 Server: Sip EXpress router (0.8.14-5 (i386/linux))
 Content-Length: 0
  
  
 --- (9 headers 0 lines)---
 Responding to challenge, registration to domain/host name budgetphone.nl
 12 headers, 0 lines
 Reliably Transmitting (no NAT) to 81.23.228.150:5060:
 REGISTER sip:budgetphone.nl SIP/2.0
 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
 From: sip:[EMAIL PROTECTED];tag=as7e56000d
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 103 REGISTER
 User-Agent: Asterisk PBX
 Authorization: Digest username=31717110342, realm=budgetphone.nl,
 algorithm=MD5, uri=sip:budgetphone.nl,
 nonce=42d15009299d7652e8da589cee2723af4b6a96ca,
 response=cd69279e6a2512fd48d267ceea3394da, opaque=
 Expires: 120
 Contact: sip:[EMAIL PROTECTED]
 Event: registration
 Content-Length: 0
  
  
 ---
 server*CLI
 -- SIP read from 81.23.228.150:5060:
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
 From: sip:[EMAIL PROTECTED];tag=as7e56000d
 To:
 sip:[EMAIL PROTECTED];tag=9b5971f23d18872ff678d4e9dae023f8.98b0
 Call-ID: [EMAIL PROTECTED]
 CSeq: 103 REGISTER
 Contact: sip:[EMAIL PROTECTED]:5060;q=0.00;expires=120
 Server: Sip EXpress router (0.8.14-5 (i386/linux))
 Content-Length: 0
  
  
 --- (9 headers 0 lines)---
 Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound
 Registration: Expiry for budgetphone.nl is 120 sec (Scheduling
 reregistration in 105000 ms)
 Destroying call '[EMAIL PROTECTED]'
 server*CLI
 -- SIP read from 81.23.228.150:5060:
 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
 Max-Forwards: 10
 Record-Route: sip:[EMAIL PROTECTED];ftag=as47419911;lr=on
 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0
 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa
 From: 0031172651375
 sip:[EMAIL PROTECTED];tag=as47419911
 To: sip:[EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX
 Date: Sun, 10 Jul 2005 16:37:54 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Content-Type: application/sdp
 Content-Length: 345
  
 v=0
 o=root 26318 26318 IN IP4 212.203.28.2
 s=session
 c=IN IP4 81.23.228.139
 t=0 0
 m=audio 36634 RTP/AVP 3 18 5 0 97 110 101
 a=rtpmap:3 GSM/8000
 a=rtpmap:18 G729/8000
 a=rtpmap:5 DVI4/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:97 iLBC/8000
 a=rtpmap:110 speex/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
  
 --- (15 headers 15 lines)---
 Using INVITE request as basis request -
 [EMAIL PROTECTED]
 Sending to 81.23.228.150 : 5060 (NAT)
 Found peer '31717110342'
 Reliably Transmitting (NAT) to 81.23.228.150:5060:
 SIP/2.0 407

Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

[Julian J. M.]

Try using insecure=very in your peer definition. That makes asterisk
not require authentication from your peer if comes from the ip address
give in host=xxx.xxx.xxx.xx directive.

That helped me receiving calls from my sip provider, which had exactly
the same problem.

Julian.

On 7/10/05, Peter Raaijmaker [EMAIL PROTECTED] wrote:
 (this time with subject)
 
 Hello,
 
 I'm trying to get Asterisk to accept incoming calls from budgetphone.nl.
 When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
 tone.
 I tried X-lite, which worked perfect, so my modem (with nat) probably is not
 the problem.
 I did a sip debug and got the following output.
 Because I'm new to Asterisk I can't get the error why this is not working.
 To me it all looks fine, no warnings or what so ever…
 
 The settings in sip.conf and extensions.conf are identical to those of
 http://www.voip-info.org/tiki-index.php?page=Talkin2ya
 
 Does anyone know what I'm doing wrong
 
 Thanks,
 Peter.

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RE: [Asterisk-Users] Incoming calls from BudgetPhone.nl

[Peter Raaijmaker]

Julian,

Thanks for your suggestion.
The insecure option was already in.
Atleast almost; I misspelled it.
I wrote insecurity !
Oooops!!

Thanks all!

Peter Raaijmakers.

-Oorspronkelijk bericht-
Van: Julian J. M. [mailto:[EMAIL PROTECTED] 
Verzonden: zondag 10 juli 2005 20:20
Aan: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

Try using insecure=very in your peer definition. That makes asterisk
not require authentication from your peer if comes from the ip address
give in host=xxx.xxx.xxx.xx directive.

That helped me receiving calls from my sip provider, which had exactly
the same problem.

Julian.

On 7/10/05, Peter Raaijmaker [EMAIL PROTECTED] wrote:
 (this time with subject)
 
 Hello,
 
 I'm trying to get Asterisk to accept incoming calls from budgetphone.nl.
 When I dial my budgetphone nr on a PSTN KPN line it immediately gives a
busy
 tone.
 I tried X-lite, which worked perfect, so my modem (with nat) probably is
not
 the problem.
 I did a sip debug and got the following output.
 Because I'm new to Asterisk I can't get the error why this is not working.
 To me it all looks fine, no warnings or what so ever.
 
 The settings in sip.conf and extensions.conf are identical to those of
 http://www.voip-info.org/tiki-index.php?page=Talkin2ya
 
 Does anyone know what I'm doing wrong
 
 Thanks,
 Peter.


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Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

[Rene Kluwen]

Same here,

Audio quality is ok. SIP registration sucks. The helpdesk makes me believe
that it is a problem on my side: Your asterisk doesn't respond to a sip
request in time. But I have no problems with any other provider, except
with Budgetphone. I am not even getting a SIP request, so how do I respond
to it?

Rene Kluwen
Chimit

 On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote:
 (this time with subject)

 Hello,

 I?m trying to get Asterisk to accept incoming calls from budgetphone.nl.
 When I dial my budgetphone nr on a PSTN KPN line it immediately gives a
 busy
 tone.
 I tried X-lite, which worked perfect, so my modem (with nat) probably is
 not
 the problem.
 I did a sip debug and got the following output.
 Because I?m new to Asterisk I can?t get the error why this is not
 working.
 To me it all looks fine, no warnings or what so ever?
 ?
 The settings in sip.conf and extensions.conf are identical to those of
 http://www.voip-info.org/tiki-index.php?page=Talkin2ya
 ?
 Does anyone know what I?m doing wrong
 ?

 Can you show us the relevant part in sip.conf and
 extensions.conf. It is working fine here (cept for audio
 quality and stability of the sip registration, I'm trashing
 them soon)
 If you post it I can compare it with my setup and maybe that
 will show us what's going wrong on your setup
 --
 Michiel van Baak
 http://michiel.vanbaak.info
 [EMAIL PROTECTED]
 GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

 Why is it drug addicts and computer afficionados are both called users?
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Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

[Michiel van Baak]

On 20:42, Sun 10 Jul 05, Rene Kluwen wrote:
 Same here,
 
 Audio quality is ok. SIP registration sucks. The helpdesk makes me believe
 that it is a problem on my side: Your asterisk doesn't respond to a sip
 request in time. But I have no problems with any other provider, except
 with Budgetphone. I am not even getting a SIP request, so how do I respond
 to it?
 
 Rene Kluwen
 Chimit

Yeah, I tried to explain it to them too several times.
They tell me the same story as they told you, and me too
doesn't see any requests.
The audio is ok on 80% of the calls, the other 20% is
plain horrible. Delays, echo, disconnect in middle of call.
I don't have those issues with nikotel or iax2 provider.
I'm now waiting for my new phone# with iax provider and when that
is done I will remove the budgetphone config from my system.
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?
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[Asterisk-Users] Incoming Calls

[David Sampson]

I have 2 4-port Digium FXS cards in my system. I would like
to play a different recording based on which trunk rings. Any pointers?

Thanks








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Re: [Asterisk-Users] Incoming Calls

[Carlos Chavez]

On Mon, 2005-06-06 at 15:25 -0400, David Sampson wrote:
 I have 2 4-port Digium FXS cards in my system.  I would like to play a
 different recording based on which trunk rings.  Any pointers?
 
 Thanks
 
  
This is really a no brainer if you read the documentation.  Simple have
each channel go to a different context in your dialplan where you can
tell it to use a different message.

-- 
Carlos Chvez
Director de Tecnologa
Telecomunicaciones Abiertas de Mxico
Tel: +52-55-91169161 Ext 2001

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[Asterisk-Users] Incoming calls picked-up then simply hanged-up

[fhunter]

I don't think my first posting went thru.

I am trying to set up Asterisk for the first time. I am new to this.
I am using [EMAIL PROTECTED]
I have a TDM400P with one FXO and one  FXS
 
The system is working for outgoing calls and if I test incoming calls using
.
But when doing an actual call the system seems to answer the call and then
immediately hang up.
 
I made a small test following some instructions and made changes to the
from-pstn context to look like this:
 
[from-pstn]
exten = s,1,Answer()
exten = s,2,Wait(4)
exten = s,3,Playback(goodbye)
exten = s,4,Hangup()
 
The incoming calls are set up to go from the PSTN to the Digital
Receptionist.
But I get the same behavior if I have incoming call send to the extension I
have set up.
 
Has anyone else seen this behavior? Any ideas as to what I should try?
 
Thanks in advance.
 



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[Asterisk-Users] Incoming calls picked-up then simply hanged-up

[John covici]

You should put your asterisk into verbose mode using asterisk -c
or if you are using a server asterisk -r and you can trace out
what happens and it will be in the log file called full in the
/var/log/asterisk directory and then you can probably figure out what
happened.  Your incoming call context must eventually dial an
extension, but I am sure you will see what is going on if you debug
like that.

on Thursday 05/12/2005 fhunter([EMAIL PROTECTED]) wrote
  I don't think my first posting went thru.
  
  I am trying to set up Asterisk for the first time. I am new to this.
  I am using [EMAIL PROTECTED]
  I have a TDM400P with one FXO and one  FXS
   
  The system is working for outgoing calls and if I test incoming calls using
  .
  But when doing an actual call the system seems to answer the call and then
  immediately hang up.
   
  I made a small test following some instructions and made changes to the
  from-pstn context to look like this:
   
  [from-pstn]
  exten = s,1,Answer()
  exten = s,2,Wait(4)
  exten = s,3,Playback(goodbye)
  exten = s,4,Hangup()
   
  The incoming calls are set up to go from the PSTN to the Digital
  Receptionist.
  But I get the same behavior if I have incoming call send to the extension I
  have set up.
   
  Has anyone else seen this behavior? Any ideas as to what I should try?
   
  Thanks in advance.
   
  
  
  
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RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

[fhunter]

Thanks I will give that a try.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Thursday, May 12, 2005 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Incoming calls picked-up then simply hanged-up


You should put your asterisk into verbose mode using asterisk -c or if
you are using a server asterisk -r and you can trace out what happens
and it will be in the log file called full in the /var/log/asterisk
directory and then you can probably figure out what happened.  Your incoming
call context must eventually dial an extension, but I am sure you will see
what is going on if you debug like that.

on Thursday 05/12/2005 fhunter([EMAIL PROTECTED]) wrote   I don't
think my first posting went thru.   
  I am trying to set up Asterisk for the first time. I am new to this.   I
am using [EMAIL PROTECTED]   I have a TDM400P with one FXO and one  FXS
  The system is working for outgoing calls and if I test incoming calls
using   .   But when doing an actual call the system seems to answer
the call and then   immediately hang up.
  I made a small test following some instructions and made changes to the
 from-pstn context to look like this:
  [from-pstn]
  exten = s,1,Answer()
  exten = s,2,Wait(4)
  exten = s,3,Playback(goodbye)
  exten = s,4,Hangup()
   
  The incoming calls are set up to go from the PSTN to the Digital  
Receptionist.   But I get the same behavior if I have incoming call send to
the extension I   have set up.
  Has anyone else seen this behavior? Any ideas as to what I should try?  

  Thanks in advance.
   
  
  
  
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RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

[fhunter]

This is what I got:

May 12 11:12:53 VERBOSE[1376]: -- Starting simple switch on 'Zap/4-1'

(Note that the line went dead on the calling phone before this next stuff
ever appeared)

May 12 11:13:01 WARNING[1376]: CallerID returned with error on channel
'Zap/4-1'
May 12 11:13:01 VERBOSE[1376]: -- Executing
[1;36;40mAnswer[0;37;40m([1;35;40mZap/4-1[0;37;40m, [1;35;40m[0;37;40m)
in new stack
May 12 11:13:01 DEBUG[1376]: Took Zap/4-1 off hook
May 12 11:13:01 DEBUG[1376]: Enabled echo cancellation on channel 4
May 12 11:13:01 DEBUG[1376]: Engaged echo training on channel 4
May 12 11:13:01 VERBOSE[1376]: -- Executing
[1;36;40mWait[0;37;40m([1;35;40mZap/4-1[0;37;40m, [1;35;40m5[0;37;40m)
in new stack
May 12 11:13:06 VERBOSE[1376]: -- Executing
[1;36;40mPlayback[0;37;40m([1;35;40mZap/4-1[0;37;40m,
[1;35;40mgoodbye[0;37;40m) in new stack
May 12 11:13:06 DEBUG[1376]: Scheduling timer at 160 sample intervals
May 12 11:13:06 VERBOSE[1376]: -- Playing 'goodbye' (language 'en')
May 12 11:13:07 DEBUG[1376]: Scheduling timer at 0 sample intervals
May 12 11:13:07 DEBUG[1376]: Scheduling timer at 0 sample intervals
May 12 11:13:07 VERBOSE[1376]: -- Executing
[1;36;40mHangup[0;37;40m([1;35;40mZap/4-1[0;37;40m,
) in new stack
May 12 11:13:07 VERBOSE[1376]: == Spawn extension (from-pstn, s, 4) exited
non-zero on 'Zap/4-1'
May 12 11:13:07 DEBUG[1376]: cdr_mysql: inserting a CDR record.
May 12 11:13:07 DEBUG[1376]: cdr_mysql: SQL command as follows: INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-05-12
11:13:01','\Francine Walker\ 201','201','s','from-pstn',
'Zap/4-1','','Hangup','',6,6,'ANSWERED',3,'')
May 12 11:13:07 DEBUG[1376]: Hangup: channel: 4 index = 0, normal = 18,
callwait = -1, thirdcall = -1
May 12 11:13:07 DEBUG[1376]: disabled echo cancellation on channel 4
May 12 11:13:07 DEBUG[1376]: Set option TDD MODE, value: OFF(0) on Zap/4-1
May 12 11:13:07 DEBUG[1376]: Updated conferencing on 4, with 0 conference
users
May 12 11:13:07 VERBOSE[1376]: -- Hungup 'Zap/4-1'
May 12 11:13:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:13:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:15:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:15:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:17:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:17:26 DEBUG[1376]: Manager received command 'Command'
May 12 11:17:26 DEBUG[1376]: Manager received command 'Command'


Any ideas, I don't see any errors reported. Just that callerID warning.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Thursday, May 12, 2005 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Incoming calls picked-up then simply hanged-up


You should put your asterisk into verbose mode using asterisk -c or if
you are using a server asterisk -r and you can trace out what happens
and it will be in the log file called full in the /var/log/asterisk
directory and then you can probably figure out what happened.  Your incoming
call context must eventually dial an extension, but I am sure you will see
what is going on if you debug like that.

on Thursday 05/12/2005 fhunter([EMAIL PROTECTED]) wrote   I don't
think my first posting went thru.   
  I am trying to set up Asterisk for the first time. I am new to this.   I
am using [EMAIL PROTECTED]   I have a TDM400P with one FXO and one  FXS
  The system is working for outgoing calls and if I test incoming calls
using   .   But when doing an actual call the system seems to answer
the call and then   immediately hang up.
  I made a small test following some instructions and made changes to the
 from-pstn context to look like this:
  [from-pstn]
  exten = s,1,Answer()
  exten = s,2,Wait(4)
  exten = s,3,Playback(goodbye)
  exten = s,4,Hangup()
   
  The incoming calls are set up to go from the PSTN to the Digital  
Receptionist.   But I get the same behavior if I have incoming call send to
the extension I   have set up.
  Has anyone else seen this behavior? Any ideas as to what I should try?  

  Thanks in advance.
   
  
  
  
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Re: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

[Julian J. M.]

Are you sure you have context=from-pstn in your zapata.conf for the
fxo channels?

Julian.

On 5/12/05, fhunter [EMAIL PROTECTED] wrote:
 I don't think my first posting went thru.
 
 I am trying to set up Asterisk for the first time. I am new to this.
 I am using [EMAIL PROTECTED]
 I have a TDM400P with one FXO and one  FXS
 
 The system is working for outgoing calls and if I test incoming calls using
 .
 But when doing an actual call the system seems to answer the call and then
 immediately hang up.

 The incoming calls are set up to go from the PSTN to the Digital
 Receptionist.
 But I get the same behavior if I have incoming call send to the extension I
 have set up.
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RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up

[fhunter]

Yep. Check context and it point to from-pstn

Any other ideas.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M.
Sent: Thursday, May 12, 2005 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming calls picked-up then simply hanged-up


Are you sure you have context=from-pstn in your zapata.conf for the fxo
channels?

Julian.

On 5/12/05, fhunter [EMAIL PROTECTED] wrote:
 I don't think my first posting went thru.
 
 I am trying to set up Asterisk for the first time. I am new to this. I 
 am using [EMAIL PROTECTED] I have a TDM400P with one FXO and one  FXS
 
 The system is working for outgoing calls and if I test incoming calls 
 using . But when doing an actual call the system seems to answer 
 the call and then immediately hang up.

 The incoming calls are set up to go from the PSTN to the Digital 
 Receptionist. But I get the same behavior if I have incoming call send 
 to the extension I have set up.
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