Forum posts to 'Asterisk Support'

Re: Asterisk Located in Toronto ,Pakistani user cant register

shah — Sun, 05 Jun 2011 07:13:38 +0300

5060 port by default blocked in pakistan

Change port ...................... 5060

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asterisk configuration

hashaam — Fri, 25 Feb 2011 10:07:31 +0200

hello.....i new to asterisk......i have installed asterisk in my local machine in linux.....i have also installed a virtual computer...virtualbox in this system......i than installed 2 twinkle phones....one in my local machine and other in the virtual machine......and i have registered them both with asterisk........there exists a problem which is that i am able to call from virtual machine to local machine and they both connect but when i try to call from host machine to virtual machine it does not work......
following are my sip.config and extension.config files.......here billy is in host machine and blaine is in virtual machine.....

sip.config:

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[blaine]
type=friend
context=phones
host=dynamic

[billy]
type=friend
context=phones
host=dynamic
...........................................

extension.config:

[globals]

[general]
autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]

[internal]
exten => s,1,Verbose(1|Echo test application)
exten => s,n,Echo()
exten => s,n,Hangup()

exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/billy,30)
exten => 1000,n,Hangup()

exten => 1002,1,Verbose(1|Extension 1002)
exten => 1002,n,Dial(SIP/blaine,30)
exten => 1002,n,Hangup()

[phones]
include => internal
include => default

Posted to: asterisk configuration Show Thread | Post Reply

TDM800 for sale

pakion — Mon, 11 Oct 2010 21:05:09 +0300

Hi and Sallam to all.

Dear i have one Asterisk 8 Ports TDM800 Module for sale any one interested do call me.

Pakion
92-333-2131218

Posted to: TDM800 for sale Show Thread | Post Reply

Digium Wildcard TE405P for sale

fshals — Sat, 18 Sep 2010 14:31:38 +0300

Digium Wildcard TE405P

The TE405P is the next generation of Digium hardware that improves performance and scalability through bus mastering architecture. Use the telephony card selector to identify a card that fits your requirements.

The TE405P supports both E1, T1, J1 environments and is selectable on a per-card or per-port basis. This feature enables signaling translation between E1, T1, and J1 equipment and allows inexpensive T1 channel banks to connect with E1 circuits. Because the TE405P improves I/O speed by up to 10 times, the result is reduced CPU usage and increased card density per server.

Digium has designed the TE405P to be fully compatible with existing software applications and it is fully integrated with the Asterisk Open Source PBX/IVR platform. Also, the open source driver supports an API interface for custom application development. With the combination of Digium Hardware and Asterisk software, numerous combinations of telephony configurations are possible. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications.

The TE405P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features.

The TE405P is for use only with a 5.0 volt PCI slot. The TE410P is for use only with a 3.3 volt PCI slot - typically available on newer server motherboards and in 64-bit PCI bus architectures.

AOA ,
Dear members i have this card now i want to sell it it is used card with extension slot i am using it from last year now i want to sell it if any one intrested then mail me on faisalshals@gmail.com , faisalshals@yahoo.com & hotmail.com same
Thanks regards
Syed Faisal Saleem

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Re: Asterisk Located in Toronto ,Pakistani user cant register

astpanj — Tue, 14 Sep 2010 11:27:04 +0300

What method using to connect to your asterisk server. SIP or what?

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Re: rejected because extension not found

astpanj — Tue, 14 Sep 2010 11:14:53 +0300

Sip.conf
[PHONE1]
type=friend
context=default
regexten=1234
callerid="User1" <1234>
host=dynamic
disallow=all
allow=ulaw

Context default does not exist in your extensions.conf so change it to context=local. If it does not work you may come back here.

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Re: Call get's disconnected after pressing *

Farhan H — Tue, 10 Aug 2010 19:19:50 +0300

Where are you running this server

I am located in Toronto and running asterisk server in Toronto
for some reason none of my user from Pakistan are able to connect to my server from Pakistan.People from US .Kuwait,Saudia Arabia are able to connect.

My Server is Running Asterisk 1.6 running on 5060 port

Any advise will help

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Re: rejected because extension not found

Rais Ali — Sun, 01 Aug 2010 02:33:07 +0300

Dear

It is easy to start with Web GUI of asterisk server that is FreePBX....using that you dont have to manuly configuration into file but configure it through GUI interface

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Re: Dialplan format

Rais Ali — Sun, 01 Aug 2010 02:30:38 +0300

Assalamoalikum

Dear Qasim are you using Web GUI of asterisk server ( FREEPBX)...voice mail ,IVR ,Music -on -Hold can easily be configured using FreePBX.....

try this

As far as Dialplan is concerned i do not have much experience with it i believe it define .....how call will be treated once it reached at successful destination. has nothing to do with registering extension or routing.

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Re: Asterisk Located in Toronto ,Pakistani user cant register

Farhan H — Mon, 12 Jul 2010 20:18:25 +0300

Fahmad can you please reply to this post

farhan

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Asterisk Located in Toronto ,Pakistani user cant register

Farhan H — Sun, 13 Jun 2010 19:18:24 +0300

Salam
I am located in Toronto and running asterisk server in Toronto
for some reason none of my user from Pakistan are able to connect to my server from Pakistan.People from US .Kuwait,Saudia Arabia are able to connect.

My Server is Running Asterisk 1.6 running on 5060 port

Any advise will help

Farhan

Posted to: Asterisk Located in Toronto ,Pakistani user cant register Show Thread | Post Reply

Dialplan format

qasimkhans — Tue, 16 Mar 2010 11:21:58 +0200

AOA All,
i am new in Asterisk and i am studing it. i could not understand dialplan format. how exactly i should make it. i can make outgoing calls but not understand how to config voicemail, conference calls, incoming calls,music on hold etc.... can any one please guide me how to make dialplan. a easy format so that i can do some work on asterisk.
you help will be highly appreciated.
Thanks,

Qasim.

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Re: Cannot make outbound calls

agent — Wed, 06 Jan 2010 09:30:07 +0200

i have resolved the problem! In order to make outbound calls, i have to set the callerid equal to the master number
exten => 277,n,set(callerid=xxxxxxxx)
exten => 277,n,dial(SIP/xxx@xxxxx)

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Re: Cannot make outbound calls

scalarshot — Wed, 06 Jan 2010 09:00:13 +0200

Hello, can you still facing the same prob if yes then send us some more details what hardware you are using youe dialplan config

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Cannot make outbound calls

agent — Mon, 23 Nov 2009 14:50:07 +0200

AoA, I am trying to intiate an outbound call on cell number. But gwtting hte following error, Can any one help me?

This is the related CLI mode Execution...which shows the status

Executing [s @macro-dial-rozee:5] Dial("SIP/0428642281-08656878", "SIP/03224657607@0428642281|30") in new stack
-- Called 03224657607@0428642281
-- Got SIP response 480 "No Routes Found" back from x.x.x.x( not shown)
-- SIP/0428642281-086a9ae8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

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Problem user registration

farooq — Fri, 13 Nov 2009 08:43:05 +0200

Dear All,

am having problem with user registration. when i create a user in sip.conf and register that user on x-lite the useer register successfully. but after some time user get unregister and show (unknown) or (unreachable) by it self. the problem is the user remain register on x-lite but not with asterisk.and asterisk cli shows following output

Asterisk Ready.
*CLI> restart gracefullyOct 2 20:04:28 NOTICE[2545]: chan_sip.c:11661 sip_poke_noanswer: Peer '3000' is now UNREACHABLE! Last qualify: 0
sip show peers
Name/username Host Dyn Nat ACL Port Status
3000/3000 192.168.2.250 D N 12470 UNREACHABLE
2000/2000 (Unspecified) D N 0 UNKNOWN
1000/1000 192.168.2.33 D N 54758 OK (103 ms)
3 sip peers [1 online , 2 offline]

Kindly someone help me please.
Thanks.
Regards,
Farooq.
Iphonica LLC

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System is not recording calls

Bunny — Wed, 22 Jul 2009 06:54:23 +0300

Salam

I have configured elastic 1.5 for out bound. Calls are going through but no calls recording in /var/spool/asterisk/monitor.I have configured as Always Recording

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rejected because extension not found

mogambo — Tue, 07 Jul 2009 15:57:44 +0300

I recently installed asterisk and trying to get it working. My X-Lite phones are registered but i cant make calls between two phones. I only modified sip.conf & extensions.conf do i need to modify any other any other file as well.
[root@asterisk1 asterisk]# cat /etc/asterisk/extensions.conf
[globals]
PHONE1=SIP/1234
PHONE2=SIP/1111

[macro-oneline]
exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup

[local]
exten => 1234,1,Macro(oneline,${PHONE1})
exten => 1111,1,Macro(oneline,${PHONE2})

Sip.conf
[PHONE1]
type=friend
context=default
regexten=1234
callerid="User1" <1234>
host=dynamic
disallow=all
allow=ulaw

[PHONE2]
type=friend
context=default
regexten=1111
callerid="User2" <1111>
host=dynamic
disallow=all
allow=ulaw

I am getting the following error:
[Jul 7 16:54:52] NOTICE[27657] chan_sip.c: Call from 'PHONE1' to extension '1111' rejected because extension not found.

SIP Debug:

--- SIP read from UDP://10.16.103.244:35672 --->
INVITE sip:1111@10.16.103.173 SIP/2.0
Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:PHONE1@10.16.103.244:35672>
To: "1111"<sip:1111@10.16.103.173>
From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103d stamp 53117
Content-Length: 267

v=0
o=- 6 2 IN IP4 10.16.103.244
s=CounterPath X-Lite 3.0
c=IN IP4 10.16.103.244
t=0 0
m=audio 26956 RTP/AVP 107 0 8 101
a=alt:1 1 : RbbV/Xv3 90+Yl1m3 10.16.103.244 26956
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 11 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.16.103.244 : 35672 (NAT)
Using INVITE request as basis request - ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
Found peer 'PHONE1' for 'PHONE1' from 10.16.103.244:35672
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.16.103.244:26956
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.16.103.244:26956
Looking for 1111 in default (domain 10.16.103.173)

<--- Reliably Transmitting (no NAT) to 10.16.103.244:35672 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;received=10.16.103.244;rport=35672
From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
To: "1111"<sip:1111@10.16.103.173>;tag=as1dc2c5a7
Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
[Jul 7 18:31:22] NOTICE[27657]: chan_sip.c:18160 handle_request_invite: Call from 'PHONE1' to extension '1111' rejected because ext
ension not found.
Scheduling destruction of SIP dialog 'ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP://10.16.103.244:35672 --->
ACK sip:1111@10.16.103.173 SIP/2.0
Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;rport
To: "1111"<sip:1111@10.16.103.173>;tag=as1dc2c5a7
From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.' Method: ACK

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extensions are not dialing

Bunny — Thu, 11 Jun 2009 20:04:39 +0300

Salam
When agents wants to dial any extension after dialing a number its not working.

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Re: SIP User Alias!

fahmad — Wed, 13 May 2009 11:05:27 +0300

Dear mzulqarnain,
Hello,

I think this can not be done.

Best Regards.

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SIP User Alias!

mzulqarnain — Wed, 13 May 2009 07:24:04 +0300

Aoa!

Is it possible to create sip alias of any given user in asterisk? e.g:

We have created following sip user: 92421234567, password: 1234

Now what we want that user can register with any of following username: 0092421234567 or 01192421234567 or +92421234567 etc. while using same password: 1234, We don't want to create multiple sip user in asterisk against a single account/customer.

What actually is going on that basic username: 92421234567 remain same but user can enter 00, 011 or + as prefix if entered by user to actual username before register from xten,ipphone etc.

I am not sure if we can create sip user alias in asterisk either by using static entry in sip.conf or realtime. May be we need to patch chan_sip to match particular pattern before asterisk look into realtime sip_buddies or sip.conf for user.

Any idea, suggestions or solution are welcome.

Thanks
Regards,
Muhammad Zulqarnain

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Re: Call get's disconnected after pressing *

mzulqarnain — Fri, 08 May 2009 07:34:17 +0300

Dear Muneeb!
There are two situations when we presed *
1) Established call dropped while you pressed *
2) Phone don't allow you to press * and give fast busy tone.

For first problem you have already been suggested to look into features.conf
For second type of problem you need to configure dialplan of your IP/Softphone if you are using Linksys IP Phone or ATA. In Web interface of Phone scroll to bottom of page and you will have a text field Dialplan: with some pre-defined patterns seperated by pipe sign ( | ).

You can define your required pattern there starting with * or any thing you need. For more help see user manual of Phone.

Thanks
Regards,
Muhammad Zulqarnain

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Re: Call get's disconnected after pressing *

justmuneeb — Wed, 06 May 2009 12:20:24 +0300

Didn't comment only work until the space is not there

like in that line commented section was ;disconnect but uncommented was *000

and yupp.. have to be reloaded.. u taught me

Regards,

Muneeb

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Re: Call get's disconnected after pressing *

fahmad — Wed, 06 May 2009 12:08:50 +0300

Dear Muneeb,
Hello,

I think if its commented then it will not going to work until or unless someone uncomment it and reload asterisk.

Best Regards.

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Call get's disconnected after pressing *

justmuneeb — Wed, 06 May 2009 12:04:30 +0300

To avoid this:

Go to cd/etc/asterisk
vi features.conf

in feature map section there would be a line similar to this:

;disconnect => *0 ; Disconnect

make it :

;disconnect => *000 ; Disconnect

Regards,

Muneeb Iqbal

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