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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: It's great to see this forum

Rais Ali — Sun, 01 Aug 2010 02:15:20 +0300

Hii

I am BE student of communication system engineering new to world of asterisk....

up till now i have successfully installed asterisk server and freepbx 2.7.My call center have following features

1) voice mail
2) IVR

using soft phone(x-lite)

and some more but I want to make a conference through my server.There will be multiple conferencing rooms users can get a or more conference room booked and enjoy conferencing

The problem is that i need road map how and where to start ... which technical skills will be required i.e. php,Mysql,C# etvc

How can i make customized interface ( plug in) through which users can enjoy conference calling

For the time being i need road map

If Anybody can help me

Thanks

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Re: Asterisk@Home Handbook

Farhan H — Mon, 12 Jul 2010 20:31:07 +0300

hi fahmad can you please reply to my post
Farhan haider

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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

Re: Urdu Numbers in Asterisk

Asad — Mon, 07 Jun 2010 17:38:34 +0300

Thats gr8 news
!As i am new to asterisk so can ne 1 Plz help Me with how can i run Your this script so that i can make asterisk say the numbers in urdu language..

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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.

Posted to: Dialplan format Show Thread | Post Reply

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)

Posted to: Cannot make outbound calls Show Thread | Post Reply

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

Posted to: Problem user registration Show Thread | Post Reply

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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Looking for a talented person who has worked over Asterisk and Digium

AJ — Wed, 08 Jul 2009 11:39:30 +0300

Hi guys...I am looking for a hardcore developer who has worked over Asterisk and digum.
You guys can send me your CV @ aasifjafri@gmail.com, you can also refer your friend or good resourse.
regards.
AJ

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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

Posted to: rejected because extension not found Show Thread | Post Reply

Re: Asterisk Pakistan Website

mogambo — Tue, 07 Jul 2009 15:53:34 +0300

Gr8 guyz, you are headed in the right direction atleast.

Posted to: Asterisk Pakistan Website Show Thread | Post Reply

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.

Posted to: SIP User Alias! Show Thread | Post Reply

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

Posted to: SIP User Alias! Show Thread | Post Reply

Re: Call recording duration

shariq — Tue, 12 May 2009 12:18:19 +0300

Remember that record command also plays a beep on the voice channel when recording start.

Posted to: Call recording duration Show Thread | Post Reply

Re: Call recording duration

fahmad — Mon, 11 May 2009 18:52:40 +0300

Dear arfeen,
Hello,

Record Command will record user voice input to a file.

FYI, http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Record

Monitor Command will record a telephone conversation to a sound file.

FYI, http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor

MixMonitor Command will record A Call Natively.

FYI, http://www.voip-info.org/wiki/view/MixMonitor

Best Regards.

Posted to: Call recording duration Show Thread | Post Reply

Re: Call recording duration

arfeen — Mon, 11 May 2009 14:54:56 +0300

whats the diff between Mix Monitor and Asterisk CMD.

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Re: Call recording duration

fahmad — Mon, 11 May 2009 14:37:35 +0300

Dear arfeen,
Hello,

I think you can do this using `Record cmd`. Please check below link for more information.

FYI, http://www.voip-info.org/wiki/index.php?page=Asterisk%2Bcmd%2BRecord

Best Regards.

Posted to: Call recording duration Show Thread | Post Reply

Call recording duration

arfeen — Mon, 11 May 2009 13:36:21 +0300

is it possible to record particular duration of the call? say first 30 seconds of the call .

Posted to: Call recording duration Show Thread | Post Reply

Re: Urdu Numbers in Asterisk

mzulqarnain — Fri, 08 May 2009 18:36:51 +0300

Thats great news.

Posted to: Urdu Numbers in Asterisk Show Thread | Post Reply

Re: Urdu Numbers in Asterisk

fahmad — Fri, 08 May 2009 17:36:08 +0300

Dear Nasir,
Hello,

Yes, i have seen that it has been commit into SVN Trunk. Which means it will be available into next stable release and available for any one willing to use latest SVN Trunk.

Best Regards.

Posted to: Urdu Numbers in Asterisk Show Thread | Post Reply

Re: Urdu Numbers in Asterisk

nasirq — Fri, 08 May 2009 08:40:46 +0300

I am not sure how asterisk version releases work, but I believe it has been added to SVN, and if you download the latest code via SVN, you will get the patched files.

Regarding, 1.4, i actually made it on 1.4. There is one function, its declaration and one addition in the ifelse area to use this function. So it should be pretty easy to put it into 1.4.

Still if you face any problem, send me the main/say.c file, and I'll send it to you patched.

I cannot a generic patch for all 1.4 versions as this file is very frequently changed as people add more languages to it.

Posted to: Urdu Numbers in Asterisk Show Thread | Post Reply

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: Urdu Numbers in Asterisk

mzulqarnain — Fri, 08 May 2009 07:24:13 +0300

Dear Nasir,
Great Work! Congratulations to you to take this first step.

When should we expect this patch to be commited and avaible in next release of asterisk 1.6.

What about Patch release for Asterisk 1.4

Thanks
Regards,
Muhammad Zulqarnain

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Re: Urdu Numbers in Asterisk

fahmad — Thu, 07 May 2009 17:47:00 +0300

Dear Nasir,
Hello,

It is great that you have made this patch. I hope many people will get advantage from this patch.

Best Regards.

Posted to: Urdu Numbers in Asterisk Show Thread | Post Reply

Urdu Numbers in Asterisk

nasirq — Thu, 07 May 2009 17:27:17 +0300

I have made a patch and uploaded the related voice files for speaking numbers in Urdu IVR.

Please see http://bugs.digium.com/view.php?id=15034

Posted to: Urdu Numbers in Asterisk Show Thread | Post Reply

Asterisk Training in Karachi

fear999 — Thu, 07 May 2009 09:56:46 +0300

Emergen consulting is pleased to announce that it would be holding a 5
day Asterisk bootcamp in Karachi. This hands-on training session,
designed for people with no prior Linux or Asterisk experience, will get
you started quickly and efficiently with the fastest growing telephony
platform.

What: 5-Day Asterisk Training by Emergen Consulting

When: 1st June - 5th June

Where: Karachi, Pakistan.

Note: Get an early bird discount of PKR 2,000.00 on registering before 10th May 2009.

For details please visit our website at:

http://www.emergen.biz/services/training

Alternatively for more information you can either reply to this email or
send us a query at:

trainings@emergen.biz

Or call us at :

021.529.0381-3

Posted to: Asterisk Training in Karachi Show Thread | Post Reply

Re: It's great to see this forum

fahmad — Thu, 07 May 2009 07:48:05 +0300

Dear Geoff,
Hello,

Welcome to the forum. I hope you will be enjoy your time here by helping and sharing your experiences on Asterisk.

Best Regards.

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Re: Asterisk Pakistan Website

mzulqarnain — Thu, 07 May 2009 06:36:18 +0300

Thanks Buddy for your effort and Time.
Regards,
Muhammad Zulqarnain

Posted to: Asterisk Pakistan Website Show Thread | Post Reply

It's great to see this forum

Geoff — Wed, 06 May 2009 19:42:41 +0300

My name is Geoff Love and I'm a sales engineer for Teliax. We work very closely with the Asterisk community.

https://teliax.com/?referral_code=11

About Us

Teliax, Inc is a global leader in voice and data services. Built from the network up, Teliax is managed by a team of technologists that embrace the benefits of open source software and cutting edge network architecture. Teliax is a privately held company located at the base of the Rocky Mountains in Denver, Colorado. The Teliax network is fully redundant, multi homed and distributed across multiple POP’s in Denver, New York, Atlanta and Los Angeles. Teliax offers flexible plans and a scalable solution for residential, small to medium size business and enterprise/wholesale markets.

Please let me know if we can assist you. We also have a great wholesale platform.

Thank,

Geoff Love
303-629-8304
GLove@Teliax.com

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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.

Posted to: Call get's disconnected after pressing * Show Thread | Post Reply

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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[Packtpub Book] Highly recommended first Asterisk Book

fahmad — Wed, 06 May 2009 11:45:33 +0300

Building Telephony Systems with Asterisk

to buy:
http://www.packtpub.com/asterisk/book

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Asterisk@Home Handbook

fahmad — Wed, 06 May 2009 11:43:35 +0300

- Old Link:

http://asteriskathome.sourceforge.net/handbook/

- New Link:

http://www.voip-info.org/wiki/view/Asterisk%40home+Handbook+Wiki

Posted to: Asterisk@Home Handbook Show Thread | Post Reply

[O'Reilly Book] Asterisk - The Future of Telephony

fahmad — Wed, 06 May 2009 11:40:16 +0300

I recomened every one to get their copy of Asterisk - The Future of Telephony. Created by the folks behind the Asterisk Documentation Project.

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Asterisk Pakistan Website

fahmad — Wed, 06 May 2009 11:29:01 +0300

Dear Members,
Hello,

We have registered asteriskpakistan.com on 2nd May 2009 and site is operational and working on 6th May 2009.

Best Regards.

Posted to: Asterisk Pakistan Website Show Thread | Post Reply