Re: [asterisk-users] Digium or Sangoma? What happened to Digium cards

[bilal ghayyad]

Hello;
I need to buy one analoge card of 8 fxo ports of pci express type and to 
support echo cancellation.
So, if no more digium, and if you do not recommend sangoma, so what is the 
solution? Which brand I should buy?
RegardsBilal-‐-‐--
I'm disappointed with Sangoma!

I have one of those Digium S101i (iaxy) adapters that is still working (in 
production), doesn't need any drivers.

After, I purchased Sangoma USBfxo (U100) adapter, never had a chance to use it 
(still brand new in a box) that require some extra driver to run it, driver 
that is no longer available.  And the U100 is obsolete.
So my advise stay away from Sangoma

Joseph
On 1/12/21 4:17 PM, John Kiniston wrote:
> Sangoma purchased Digium.
> 
> You can find Sangoma cards at https://www.sangoma.com/telephony-cards/
> 
> On Tue, Jan 12, 2021 at 2:29 PM bilal ghayyad  wrote:
> 
>> Hello All;
>>
>> We were using Digium cards, now I am not able to reach for digium website
>> that contains the telephony cards and Asterisk website currently is taking
>> us for Sangoma, so what happened in Digium cards?
>>
>> Regards
>> Bilal  
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[asterisk-users] Digium or Sangoma? What happened to Digium cards

[bilal ghayyad]

Hello All;
We were using Digium cards, now I am not able to reach for digium website that 
contains the telephony cards and Asterisk website currently is taking us for 
Sangoma, so what happened in Digium cards?
RegardsBilal-- 
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[asterisk-users] 10 Caller IDs to be used randomly or progressively

[bilal ghayyad]

Hello;
I have 10 Caller IDs and I need each call (each time) to use one of these 
Caller IDs to be the caller id.
I know that I can use this syntax as example:
exten => _90ZXX,1,Set(CALLERID(num)=01747576)

But how I can set the callerid each time from be one of the 10 caller ids that 
are allowed for me?For example: first call to use caller id 01747576 and second 
call to use caller id 01747577 and third call to use caller id 01747578 and so 
on, how?
RegardsBilal-- 
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Re: [asterisk-users] asterisk-users Digest, Vol 181, Issue 3

[bilal ghayyad]

 Thank you a lot for your kindly help and reply. Actually it helped me a lot.I 
was using _X. in the extensions.conf at the trunkinbound context.Can you advise 
me what is the difference between _X. and s? In other words, when it is better 
to use s and when it is better to use _X.?
Again, I am fully thanks for you.RegardsBilal
> Hello;
> 
> I am facing a trouble with the SIP service provider, they are saying 
> that there is a problem related to message option 200 (the heartbeat), 
> so what is required to add this for the sip configuration? Below is my 
> sip debug trace log with the them and the sip peer configuration:

OPTIONS is treated as if it were an INVITE, so it looks up the extension in the 
dialplan. The following shows what extension and context:

[Sep 4 12:42:20] Looking for s in trunkinbound (domain 10.240.147.26)

If you add an "s" extension to the "trunkinbound" context it should then 
respond 200 OK.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org  -- 
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[asterisk-users] SIP trunk problem: Message option 200 (heartbeat)

[bilal ghayyad]

Hello;
I am facing a trouble with the SIP service provider, they are saying that there 
is a problem related to message option 200 (the heartbeat), so what is required 
to add this for the sip configuration? Below is my sip debug trace log with the 
them and the sip peer configuration:


[Sep  4 12:42:18]<>

[Sep  4 12:42:18]Scheduling destruction of SIP 
dialog'kntaydtnxawwiyky4yaadanied4lk4ys@139.110.215.10' in 32000 ms (Method: 
OPTIONS)

[Sep  4 12:42:19]Really destroying SIP dialog 
'xidaktlblveesdlbtkbyyexelaiibtvx@139.110.215.10'Method: OPTIONS

[Sep  4 12:42:20]

[Sep  4 12:42:20] <--- SIP read fromUDP:10.215.110.139:5060 --->

[Sep  4 12:42:20] OPTIONS sip:10.240.147.26:5060SIP/2.0

[Sep  4 12:42:20] Via: 
SIP/2.0/UDP10.215.110.139:5060;branch=z9hG4bKxlaisbketikainbbedltysdla;Role=3;Hpt=8e78_16;TRC=-;pth=0;X-HwDim=4

[Sep  4 12:42:20] Call-ID:aecdkcavavdticeydtswaewesttbbad4@139.110.215.10

[Sep  4 12:42:20] From:;tag=yaekiyny

[Sep  4 12:42:20] To: 

[Sep  4 12:42:20]CSeq: 1 OPTIONS

[Sep  4 
12:42:20]Contact:;expires=65535

[Sep  4 12:42:20]Accept: application/sdp

[Sep  4 12:42:20]Max-Forwards: 70

[Sep  4 12:42:20]Content-Length: 0

[Sep  4 12:42:20]

[Sep  4 12:42:20] <->

[Sep  4 12:42:20] --- (10 headers 0 lines) ---

[Sep  4 12:42:20] Sending to 10.215.110.139:5060(NAT)

[Sep  4 12:42:20]Looking for s in trunkinbound (domain 10.240.147.26)

[Sep  4 12:42:20]

[Sep  4 12:42:20] <--- Transmitting (NAT) to10.215.110.139:5060 --->

[Sep  4 12:42:20] SIP/2.0 404 Not Found

[Sep  4 12:42:20] Via: 
SIP/2.0/UDP10.215.110.139:5060;branch=z9hG4bKxlaisbketikainbbedltysdla;Role=3;Hpt=8e78_16;TRC=-;pth=0;X-HwDim=4;received=10.215.110.139;rport=5060

[Sep  4 12:42:20] From: ;tag=yaekiyny

[Sep  4 12:42:20] To:;tag=as7efcc39d

[Sep  4 12:42:20]Call-ID: aecdkcavavdticeydtswaewesttbbad4@139.110.215.10

[Sep  4 12:42:20]CSeq: 1 OPTIONS

[Sep  4 12:42:20]Server: Asterisk PBX 13.24.1-vici

[Sep  4 12:42:20]Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY, INFO,PUBLISH, MESSAGE

[Sep  4 12:42:20]Supported: replaces, timer

[Sep  4 12:42:20]Accept: application/sdp

[Sep  4 12:42:20]Content-Length: 0

And below is the sip peer configuration:
[ooredoo]type=friendhost=10.215.110.139bindport=5060dtmfmode=autocontext=trunkinboundcanreinvite=nodisallow=allallow=ulawallow=alawallow=g729allow=gsmtrustrpid=yesnat=force_rport,comediainsecure=invite,port

So what is needed to resolve this [Sep  4 12:42:20] SIP/2.0 404 Not Found ?What 
is needed to be added for the sip peer configuration?
RegardsBilal-- 
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Re: [asterisk-users] Sending SMS and SIM card

[bilal ghayyad]

Hello John;
And for GSM calls, u were using sip trunk from asterisk to these gateways?
And how you were sending sms?

> I use VoIP Innovations and ThinQ (formally SIPRoutes) and they both support 
> SMS. That way it’s very easy to write it into the dial plan.
RegardsBilal-- 
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Re: [asterisk-users] Sending SMS and SIM card

[bilal ghayyad]

Hello;chan_dongle can be used for sms and for gsm calls at the same time, how?
Any small example how to send gsm calls through chan_dognle and how to send sms 
through chan_dongle?

> You can use a cheap 3G-USB-dongle and chan_dongle.  
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Re: [asterisk-users] Sending SMS and SIM card

[bilal ghayyad]

 Thank you Steve.Regarding to Goip32 that you used it before: how you were 
handling the received messages?In other words: if you sent a message for 
someone and he replied for you, how you were able to see the reply? And was it 
possible to have any action based on his reply (for example, forward the 
message to email)?
Also regarding to the Goip32: how you were sending the SMS messages? From CRM 
connected to it or it was having a web interface for sending the SMS messages?
Was you able to use Goip32 for GSM voice calls (sending and receiving)?
RegardsBilal
> Is it possible to send SMS from asterisk? Using DAHDI or using what is 
> possible?

You can use an SMS provider like Twillio.


> And, is there a card that can be fixed in the machine and insert the SIM 
> card in this card to be used for GSM calls and sending SMS through 
> asterisk? Through which channel? Is it DAHDI or something else?

I've never used an internal card, but what you're looking for is a GSM 
gateway.

I used a Goip32 (32 SIMs, 32 channels) a couple of years ago. It is an 
external box you hang on your network via Ethernet.


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[asterisk-users] Sending SMS and SIM card

[bilal ghayyad]

Hello;
Is it possible to send SMS from asterisk? Using DAHDI or using what is possible?
And, is there a card that can be fixed in the machine and insert the SIM card 
in this card to be used for GSM calls and sending SMS through asterisk? Through 
which channel? Is it DAHDI or something else?
RegardsBilal-- 
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[asterisk-users] Button for call forward and button for pickup call of another extension

[bilal ghayyad]

Hello;
I do not know if the following feature is depending on the phone (can be 
configured on the phone it self) or need to be configured from asterisk itself:
Is it possible to configure general SIP Phone to have one button that can be 
used as following:
By pressing on it and then entering another phone extension, then all incoming 
calls for this phone will be forwarded for that extension? If someone can 
direct me how to do it in Polycom, then it is good.
And another button, by pressing on it, I can enter the extension that I need to 
pickup the call that is ringing at it. How?
I do not know if these features need to be configured from the Phone it self or 
need to be configured from asterisk and both?
RegardsBilal-- 
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[asterisk-users] Busy indicator for FXO line or extension

[bilal ghayyad]

Hello;
Is it possible to configure one button on the IP Phone (like Polycom or general 
SIP Phone) to indicate (at the phone display) that the line (the line that is 
connected for FXO port) is busy or not? If it is not busy, the user can press 
on the button to place outside call.
Also, is it possible to to configure another button to indicate if the 
extension is busy or not?
RegardsBilal-- 
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[asterisk-users] GSM card or GSM adaptor?

[bilal ghayyad]

Hello;
I need to be able to send and receive voice calls through GSM network, so do I 
need GSM adaptor that will be connected to FXO port or I can use GSM card that 
can be connected to PCI or PCI-E slot in the computer and asterisk can see this 
card through dahdi channel?I am afraid that if I used GSM card (PCI or PCI-E) 
then I need special channels other than dahdi to be installed to be able to use 
it for calls, and I do not know in this case if the channels will be configured 
as dahdi channels or what?
RegardsBilal

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[asterisk-users] sip trunk with social media

[bilal ghayyad]

Hello
It will be amazing if possible to do sip trunk with any of social media 
providers like: whatsapp, facebook, imo, viber, ... etc.Did anyone has luck 
with this? RegardsBilal

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[asterisk-users] atcom card: how it is?

[bilal ghayyad]

Hello;
I am thinking to use atcom card which can be shown in this link:AXE2G4AN - GSM 
card - Atcom_Ip phone,IP PBX,Asterisk Cards,Voip Products Manufacturer

  
|  
|   
|   
|   ||

   |

  |
|  
|   |  
AXE2G4AN - GSM card - Atcom_Ip phone,IP PBX,Asterisk Cards,Voip Products Ma...
 ATCOM is the leading VoIP hardware manufacturer in global market. We have been 
keeping innovating with customer’...  |   |

  |

  |

 

But I am afraid, because I used to use digium and I am afraid of the quality. 
Maybe someone will ask me why not to use digium? The answer: because I need the 
card to has one GSM sim port and 1 FXO port and did not find this with digium 
or sangoma. But I am afraid from ATCOM that it might be low quality.Also, I 
need to know how the quality will be in case there is GSM and FXO at the same 
card, will there be a noise or distortion?
Appreciate the kindly help and advise.RegardsBilal-- 
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[asterisk-users] SIP trunk with whatsapp

[bilal ghayyad]

Hello;
Does anyone has information if possible to setup SIP trunk with whatsapp? How 
can we let asterisk send and receive calls from whatsapp?
RegardsBilal-- 
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[asterisk-users] 488 Not acceptable here

[bilal ghayyad]

Hello List;
I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I 
am getting the following debug, can someone advise me about the solution:
<--- SIP read from Provider_IP_Address:5083 --->INVITE 
sip:22021782@Asterisk_IP_Address:5060 SIP/2.0 Via: SIP/2.0/UDP 
Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1 From: "1828444" 
;tag=rrZpHF51Z7a6D To: 
 Call-ID: 
6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5 
CSeq: 1 INVITE Max-Forwards: 68 Supported: timer Unsupported: refer Allow: 
INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY Contact: 
 Content-Length: 729 
Content-Type: application/sdp User-Agent: Netborder SS7 to VoIP Media Gateway 
5.1 Allow-Events: talk Accept: application/sdp Privacy: none X-IP-Info: 
10.11.11.3  v=0 o=FreeSWITCH 1453083377 1453083378 IN IP4 Provider_IP_Address 
s=FreeSWITCH c=IN IP4 Provider_IP_Address t=0 0 m=audio 28388 RTP/AVP 8 0 98 9 
99 100 18 3 102 101 13 a=rtpmap:98 AMR/8000 a=rtpmap:99 G7221/16000 a=fmtp:99 
bitrate=32000 a=rtpmap:100 G726-32/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 
mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=audio 
29684 RTP/AVP 4 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 
a=ptime:30 m=audio 21364 RTP/AVP 8 0 98 9 99 100 18 3 102 101 13 a=rtpmap:98 
AMR/8000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 
G726-32/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:101 
telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 
<->[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] --- 
(18 headers 29 lines) ---[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 
10:52:37] Sending to Provider_IP_Address : 5083 (no NAT)[Jan 18 10:52:37] 
VERBOSE[2421] logger.c: [Jan 18 10:52:37] Using INVITE request as basis request 
- 
6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5[Jan
 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found peer 
'gulfnet'[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP 
audio format 8[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found 
RTP audio format 0[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] 
Found RTP audio format 98[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 
10:52:37] Found RTP audio format 9[Jan 18 10:52:37] VERBOSE[2421] logger.c: 
[Jan 18 10:52:37] Found RTP audio format 99[Jan 18 10:52:37] VERBOSE[2421] 
logger.c: [Jan 18 10:52:37] Found RTP audio format 100[Jan 18 10:52:37] 
VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 18[Jan 18 
10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 
3[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio 
format 102[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP 
audio format 101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] 
Found RTP audio format 13[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 
10:52:37] Found unknown media description format AMR for ID 98[Jan 18 10:52:37] 
VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description 
format G7221 for ID 99[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 
10:52:37] Found audio description format G726-32 for ID 100[Jan 18 10:52:37] 
VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format iLBC 
for ID 102[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found 
audio description format telephone-event for ID 101[Jan 18 10:52:37] 
VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 4[Jan 18 
10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 
101[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio 
format 13[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found 
audio description format telephone-event for ID 101[Jan 18 10:52:37] 
VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 8[Jan 18 
10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 
0[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio 
format 98[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP 
audio format 9[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found 
RTP audio format 99[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] 
Found RTP audio format 100[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 
10:52:37] Found RTP audio format 18[Jan 18 10:52:37] VERBOSE[2421] logger.c: 
[Jan 18 10:52:37] Found RTP audio format 3[Jan 18 10:52:37] VERBOSE[2421] 
logger.c: [Jan 18 10:52:37] Found RTP audio format 102[Jan 18 10:52:37] 
VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 101[Jan 18 
10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 
13[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] 

Re: [asterisk-users] 488 Not acceptable here

[bilal ghayyad]

Hello;
Thanks a lot for your kindly reply.Actually the alaw is enabled at asterisk but 
what I got to know from the other side that they only enabled ulaw. Below is my 
asterisk sip configuration for the sip trunk. Please advise.
[user_name]type=peerhost=Provider_IP_Addressport=5083context=trunkinbounddisallow=allallow
 = ulaw,alaw,gsmcall-limit = 256  insecure = port,invitetrunkstyle = 
providertransport = udp  dtmfmode = rfc2833remoteregister = yescbcallerid = 
22021782qualify = yessrtpcapable = no
RegardsBilal 

On Wednesday, January 20, 2016 2:50 PM, A J Stiles 
<asterisk_l...@earthshod.co.uk> wrote:
 

 On Wednesday 20 Jan 2016, bilal ghayyad wrote:
> Hello List;
> I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and
> I am getting the following debug, can someone advise me about the
> solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE
> . [stuff deleted] .
> [Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37]<---
> Reliably Transmitting (no NAT) to Provider_IP_Address:5083 --->SIP/2.0 488
> Not acceptable here Via: SIP/2.0/UDP
> Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1;received=Pro
> vider_IP_Address From: "1828444" <sip:1828...@c4.gw>;tag=rrZpHF51Z7a6D To:
> <sip:22021782@Asterisk_IP_Address:5060>;tag=as5d16dbaf Call-ID:
> 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFq
> loi5 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,
> OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported:
> replaces Content-Length: 0 <>

"488 Not acceptable here" usually means that negotiation failed for want of 
any mutually-supported codec.  Make sure that you have "alaw", which is the 
native format used by the PSTN in civilised countries  (and therefore, there 
is little need to use anything else unless you know you will never want PSTN 
connectivity),  enabled at your end.


Can you run this command and post the output?  (It should all be on one line, 
but my mail client or yours may have eaten it)

$ awk '/[[]|allow/&&!/^[ \t]*;/{printf "%6d:%s\n",NR, $0}' 
/etc/asterisk/sip.conf

This will look for [section headers] in square brackets and lines containing 
"allow" (which also will catch "disallow") that are not commented out, in your 
SIP configuration, and print them out with line numbers.


-- 
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] Hearing peep for second call and special signal for caller

[bilal ghayyad]

Hello;
The the destination already have a call (talking) and someone called it, we 
need the caller to hear a tone which indicate that the destination has a call 
(busy) and the destination should hear a tone to indicates that someone is 
calling him. How can we do this? 
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[asterisk-users] Billing software: Other than A2Billing because of the problem with the analogue channels

[bilal ghayyad]

Hello;

I am facing a trouble with A2Billing when using analogue lines because the 
channels are not closing properly when dialing happen through A2Billing (it 
seems the dialing scenario including the hangup is not handled properly through 
A2Billing but I do not have control on this). But when I do dialing from 
asterisk and using analogue lines, I do not face a trouble because I can write 
the script in the extensions.conf in professional way to confirm that the 
channel is closed successfully.

Is there alternative Billing solution than A2Billing which has another working 
mechanism? How I can resolve such problem which is related to the analogue 
channels? 

Regards
Bilal-- 
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[asterisk-users] Alternative billing for A2Billing because of using Dial function with analogue lines

[bilal ghayyad]

Hello All;

After trying A2Billing and certainly when the trunk is analogue lines (FXO 
ports), I faced a problem that the channels were not hanged up properly from 
time to time which cause us to do restart for the dahdi. Without A2Billing, I 
was able to handle the Dial scenario properly and no hanging for the analogue 
channels and no need to restart dahdi from time to time. 

Really I would if there is alternative Billing software (open source) for 
A2Billing and its working mechanism differs than A2Billing (I always prefer to 
keep handle the Dial scenario from asterisk configuration file and not through 
the billing software to be sure that it is hanged up properly in all the 
cases), I would if there is a billing software that do only the billing part 
and send control the disconnection for the call by sending commands to asterisk 
while the dialing scenario is handled totally from the asterisk configuration.

Or at least, is there billing software that work in better performance than 
A2Billing and does not cause for me problems if the trunk is analogue?

Regards
Bilal-- 
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[asterisk-users] Using asterisk as voicemail for cisco call manager

[bilal ghayyad]

Hello;

Instead of using Cisco Unity as voicemail for Cisco Call Manager, I need to use 
asterisk to be the voicemail for the Cisco Call Manager version 7 which 
supports SIP.

Did anyone try this? Was it a successful implementation?

If yes, I hope that someone gives a steps to help me.

Regards
Bilal-- 
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[asterisk-users] Polycom does not register from outside to asterisk

[bilal ghayyad]

Hello;

I have asterisk Asterisk 1.8.23.0-vici and Polycom 331 and I am able to 
register from local area network and not able to register from outside the 
office. Also from outside the office, I am able to register via PhonerLite 
softphone and not able to register via Zoiper softphone.

So from outside the office, I am not able to register from Zoiper softphone and 
not able to register from Polycom 331. 

I set the externip to the router real IP address.
Also, I set nat=yes.

What could be the problem? Why I am able to register from outside the office 
using Phoner Lite softphone and not able using zoiper softphone or Polycom 331?

Is there anything special settings that can resolve this problem?

Regards
Bilal-- 
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[asterisk-users] Integration with outlook

[bilal ghayyad]

Hello;

Is there a method way to be able to dial the phone number through asterisk 
from the outlook email contact?

Regards
Bilal-- 
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Re: [asterisk-users] Maximum number of users

[bilal ghayyad]

Thanks a lot for the help from the all.

Without using PBX functionalities (like conference or pickup and so on), only 
basic calls. So how many concurrent calls can support?

The idea is, we need to use Asterisk with ISP which will be service provider 
for sip calls for the subscribers, and the asterisk should be connected with 
E1s to do calls within the country.

The registered users will reach up to 100 000 users and the concurrent calls 
will reach up to 2000 or 3000 calls.

Appreciate the kindly advise.
Regards
Bilal



On Wednesday, December 18, 2013 6:10 PM, Tech Support 
aster...@voipbusiness.us wrote:
 
You can have tens of thousands of phones as long as no one makes or receives 
any calls J. The better question to ask is how many concurrent calls have 
people been able to make. The quick answer is it depends on many things. 
John 
 
From:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Wednesday, December 18, 2013 9:46 AM
To: Asterisk Users Mailing List - Non- Commercial Discussion
Subject: [asterisk-users] Maximum number of users
 
Hello;
 
Can someone advise me what is the maximum number of users (IP Phones) that can 
be supported by asterisk 1.8 or later?
 
Regards
Bilal-- 
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[asterisk-users] Maximum number of users

[bilal ghayyad]

Hello;

Can someone advise me what is the maximum number of users (IP Phones) that can 
be supported by asterisk 1.8 or later?

Regards
Bilal-- 
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[asterisk-users] Voicemail interface

[bilal ghayyad]

Hello;

Is there Interface (web based interface) that I can login as admin, check the 
emails and see the numbers that leaved voicemail and if possible to hear the 
voice message, ... etc?

Regards
Bilal-- 
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[asterisk-users] Calls Recording Solution

[bilal ghayyad]

Hello;

I am looking for calls recording solution to do recording based on the network 
traffic .. The solution to be competitive and appreciate if it is open source 
.. Any suggested one?

Regards
Bilal-- 
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Re: [asterisk-users] Calls Recording Solution

[bilal ghayyad]

Using Orecx, I can do search based on the extension or caller number or the 
time or the agent login or the mix of these fields?


Regards
Bilal



On Tuesday, October 22, 2013 6:03 AM, Paul Belanger 
paul.belan...@polybeacon.com wrote:
 
On 13-10-21 10:39 PM, bilal ghayyad wrote:

 Hello;

 I am looking for calls recording solution to do recording based on the 
 network traffic .. The solution to be competitive and appreciate if it is 
 open source .. Any suggested one?

http://www.orecx.com/

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
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[asterisk-users] ADSL and VPN router

[bilal ghayyad]

Hello;

I am looking for ADSL that supporting VPN so we can connect to it from our 
IPhone using the VPN to be able to register at the asterisk PBX. Any 
recommended one that is doing fine with voice? Also, does it support bandwidth 
priority or shaping for the protocols?

Regards
Bilal-- 
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Re: [asterisk-users] meetme conference password and time limitation

[bilal ghayyad]

So this web-meetme applicationrequires to enable the real time in asterisk? 
Where I can find documentation about web-meetme application?

Regards
Bilal



On Tuesday, October 1, 2013 6:57 PM, Dan Austin dan_aus...@phoenix.com wrote:
 
Look at Web-MeetMe ( http://sf.net/projects/web-meetme )
If you are on Asterisk 1.6.7 or later you have access to RealTime
MeetMe conference storage, otherwise you need to use a
script and Asterisk application included with the WMM download.
 
Dan
 
 
From:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Tuesday, October 01, 2013 12:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] meetme conference password and time limitation
 
Hello;
 
We need to have admin page, so the administrator can create passwords to be 
used to join the meetme conferences and can determine the allowed time .. 
 
Well, the admin interface can be done easy (I do not know if there is something 
ready), and the password and the time limitation can be added to the database 
(or even text file), but how asterisk can use it? Do I need to use the AGI to 
read/write from database and do the meetme conference within the AGI script it 
self, or there is simpler method?
 
Regards
Bilal-- 
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[asterisk-users] meetme conference password and time limitation

[bilal ghayyad]

Hello;

We need to have admin page, so the administrator can create passwords to be 
used to join the meetme conferences and can determine the allowed time .. 

Well, the admin interface can be done easy (I do not know if there is something 
ready), and the password and the time limitation can be added to the database 
(or even text file), but how asterisk can use it? Do I need to use the AGI to 
read/write from database and do the meetme conference within the AGI script it 
self, or there is simpler method?

Regards
Bilal-- 
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Re: [asterisk-users] SIM adaptor (huwewi or other)

[bilal ghayyad]

On Wednesday, September 11, 2013 1:54 PM, longst longst...@gmail.com wrote:
 
I think GoIP gsm gateway also is a good choice 

Sent from Shitian Long


On Sep 11, 2013, at 12:29 PM, bilal ghayyad bilmar...@yahoo.com wrote:


Hello;


I am looking for SIM adaptor to be connected with Asterisk to be able to send 
and receive calls from the mobile operator and if possible the same adapter to 
be used for SMS sending and receiving.


But what if anyone called this SIM card that is connected to this adapter and 
no one relied his call, how this miss call can reach for the use at the 
asterisk PBX?


Regards
Bilal
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[asterisk-users] Polycom voicemail menu and alarm as beep with light

[bilal ghayyad]

Hello;

I am using vicidial which is using asterisk 1.8, mean while when the extension 
has voicemail, I always see the red light on the Polycom and hear the beep 
sound (toot toot) in period time. Also, I can see at the LCD an option to 
select it for accessing the voicemail  but I am facing the following problems:

1) The red light and the beep: How I can let the Phone only have the red light 
without the beep sound that keep hearing it periodically and it is bothering? 
Because I tried from the Polycom web based settings but nothing related to this 
.. Maybe, it is settings need to be from the setting file that we have to place 
it in the TFTP Server? Any advise?

2) Regarding to accessing the voicemail from the phone (this option that is 
existed at the Polycom IP Phone as long the voice mail is existed), when 
clicking on this button, it is not accessing the voicemail main menu .. How I 
can do this?

Regards
Bilal--
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[asterisk-users] SIM adaptor (huwewi or other)

[bilal ghayyad]

Hello;

I am looking for SIM adaptor to be connected with Asterisk to be able to send 
and receive calls from the mobile operator and if possible the same adapter to 
be used for SMS sending and receiving.

But what if anyone called this SIM card that is connected to this adapter and 
no one relied his call, how this miss call can reach for the use at the 
asterisk PBX?

Regards
Bilal--
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[asterisk-users] Installing asterisk and dahdi on ubuntu

[bilal ghayyad]

Hello;

I am installing asterisk and dahdi on ubuntu and I used my username bghayad to 
login for ubuntu and do the installation, actually I feel my problem is related 
to the username and permission but I am not able how to fix it, I am facing 
now mainly the following two problems:

The first one, asterisk is not starting automatically although I did sudo make 
config (for asterisk and dahdi) and the asterisk and dahdi scripts have been 
created under /etc/init.d/

The second problem, I started asterisk using asterisk -cvvv and from the CLI, I 
tried dahdi show version and dahdi show status, I am getting the following 
results:

*CLI dahdi show status
No DAHDI found. Unable to open /dev/dahdi/ctl: Permission denied
Command 'dahdi show status ' failed.

*CLI dahdi show version
Failed to open control file to get version.


Below is my ubuntu information:

bghayad@Bilal:/usr/sbin$ lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
Description:    Ubuntu 12.04.1 LTS
Release:        12.04
Codename:       precise

Regards
Bilal--
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[asterisk-users] Echo Cancellation

[bilal ghayyad]

Hello;

If our Digium Telephony Card does not support echo cancellation like 
(1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the 
echo?

Regards
Bilal
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[asterisk-users] Which is the stable version to use?

[bilal ghayyad]

Hello

I need to deploy asterisk on production and same thing for DAHDI, which version 
is recommended for this?

Regards
Bilal--
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[asterisk-users] auto answer

[bilal ghayyad]

Hello;

Is it possible to configure in the sip.conf for the Phone to be auto answer?

Regards
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Re: [asterisk-users] auto answer

[bilal ghayyad]

But this not in the sip.conf, this in the extensions.conf, right?

Regards
Bilal



 From: Yasin Suluhan ysulu...@gmail.com
To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List - 
Non-Commercial Discussion asterisk-users@lists.digium.com 
Sent: Wednesday, July 17, 2013 12:21 PM
Subject: Re: [asterisk-users] auto answer
 


Hello, 

You could use Answer-After for that. But, afaik there is no definitive 
description in the RFCs on how it is used. 

You would have to enable such features on the telephones too. And I would 
expect that different phone manufacturers would probably use different 
mechanisms to enable such an option. 

Furthermore, considering the security issues this would create i wouldn' t 
recommend taking such a path. 



On Wed, Jul 17, 2013 at 12:04 PM, bilal ghayyad bilmar...@yahoo.com wrote:

Hello;


Is it possible to configure in the sip.conf for the Phone to be auto answer?


RegardsBilal
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-- 

Best Regards.


Yasin SULUHAN
Contact Information
Mobile: +90 535 656 35 55--
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Re: [asterisk-users] auto answer

[bilal ghayyad]

So it is not at asterisk configuration?

Regards
Bilal



 From: A J Stiles asterisk_l...@earthshod.co.uk
To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List - 
Non-Commercial Discussion asterisk-users@lists.digium.com 
Sent: Wednesday, July 17, 2013 12:57 PM
Subject: Re: [asterisk-users] auto answer
 

On Wednesday 17 July 2013, bilal ghayyad wrote:
 But this not in the sip.conf, this in the extensions.conf, right?
 
 Regards
 Bilal

No.  This would be set up in the phone's own configuration file, which in turn 
depends on the make and model of phone  (and its location depends on your site 
setup).

-- 
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[asterisk-users] Digitial Phones

[bilal ghayyad]

Hello;

Does asterisk support Digital Phone devices? If yes, what is the required cards 
and in which channel to do the configuration? Is it dahdi or something else?

In other words, the customer does not need IP Phones.

Regards
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[asterisk-users] Xeon Server and total number of extensions

[bilal ghayyad]

Hello;

If I have load up to 220 extensions with 50 concurrent calls. Can one hardware 
server carry all this load? What is the hardware server required for this?

Regards
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[asterisk-users] PoE module

[bilal ghayyad]

Hello;

We have a cisco switches but they are not PoE and we need only to have PoE 
device so the cables come for it first to provide the power and then goes to 
the switch (to be like batch panel), is there something like this that can be 
used for the IP Phones?

Regards
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[asterisk-users] PoE L3 Switches

[bilal ghayyad]

Hello;

Anyone used PoE L2 network switches other than cisco and recommend this for us? 
We need it to be stable and costly effective.

Regards
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[asterisk-users] Dongle or extra channel and sip SMS

[bilal ghayyad]

Hello;

I need to be able to send SMS messages for campaign or for specific users, also 
I need to be able to receive SMS messages and do automatic reply.
Do I have to use dongle or extra channel? What is the difference?
Also, I read that there is SMS through sip, how this work and what is the 
difference between the sip SMS and gsm sip? If I need to send sip SMS, how 
destination will receive it? What is required for destination phone to receive 
this sip SMS?

Thanks for the help.

Regards
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[asterisk-users] CTI for asterisk?

[bilal ghayyad]

Hello;

Is there CTI module in asterisk with CTI client to login and logout and do 
ready and pause?

Regards 
Bilal--
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[asterisk-users] autoanswer

[bilal ghayyad]

Hello;

To let the Phone answer automatically, this can be configured from asterisk (at 
the sip.conf for the phone)? Or it has to be from the IP Phone? Because, some 
phones does not support auto answer, also we do not need to do it for each 
Phone.

Regards
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[asterisk-users] How to know the conflict in the dependencies?

[bilal ghayyad]

Hello;

When I type make menuselect and finding the channels that has the sign XXX 
before it (this at the driver), how can I know the dependencies that are 
causing this conflict?

Regards
Bilal

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[asterisk-users] Jabber

[bilal ghayyad]

Hello;

Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to communicate 
with them. But, how much jabber channel in asterisk is stable and updated?

Regards
Bilal

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[asterisk-users] Integration with skype

[bilal ghayyad]

Hello;

There is no free channel to be used to have integration between asterisk and 
skype? What is the software that I can use to send and receive chat messages on 
skype network?

Regards
Bilal

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[asterisk-users] Asterisk 1.8 vici and the fax, SMS, gtalk, Jaber channels

[bilal ghayyad]

Hello;

As I am using vicidial and its asterisk version which is 1.8, I need to know 
the required channels to be existed so the asterisk will support fax, SMS, 
gtalk, Jaber? In other words, how I can know that it is enabled in this 
asterisk (actually it is 1.8.21-vici)?

Regards 
Bilal--
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[asterisk-users] wanpipe and digium, oslec and hardware echo canceller

[bilal ghayyad]

Hello All;

Wanpipe is working only with sangoma cards so it does not work with digium 
cards?

Also, who is better: to have echo canceler built in with the hardware or using 
olsec?

Regards
Bilal

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[asterisk-users] Which channels are required for FAX, GTALK and Jaber

[bilal ghayyad]

Hello;

To be able to send and receive faxes through asterisk and to be able to have 
trunk with google voice and to be able to have integration with those that 
support Jaber .. What are the channels and libs that I have to be sure that 
they are existed?

Regards
Bilal

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[asterisk-users] Obtaining high voice quality in dahdi channel

[bilal ghayyad]

Hello;

What is the best method to let the voice quality through Dahdi channels to be 
clear and no echo? Is it the wanpipe or it is working only with sangoma?

Regards
Bilal

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[asterisk-users] SMS Scenario

[bilal ghayyad]

Hello;

I need two scenarios:

1) If someone sent SMS message, then we need to query information from the 
database based on information sent by the SMS (like the name or the mobile 
number), after querying from the database, we need to reply by the SMS. Can 
asterisk do this? To which level?

2) I have vtiger CRM, and it is possible to send SMS (there is a button: send 
SMS), the question is: what is the required to be able to send the message from 
the vtiger CRM via asterisk SMS module?

Regards
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[asterisk-users] asterisk 1.4 and SMS module

[bilal ghayyad]

Hello;

As I am using vicidial and still vicidial is using asterisk 1.4, so how is the 
SMS module with asterisk 1.4? Is it stable?


Also, I am looking to integrate with social medial like whatsapp and facebook, 
so how is asterisk 1.4 with jaber channel?

Regards
Bilal

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[asterisk-users] Asterisk 1.8 and 11

[bilal ghayyad]

Hello;

How I can compare between Asterisk 1.8 and 11 with reference to the following 
points:

1) SMS.
2) gtalk and other social media.
3) GUI.
4) Any main difference?

Regards
Bilal

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[asterisk-users] To enhance the voice quality of the SIP trunk

[bilal ghayyad]

Hello;

I have a SIP trunk with a service provider, the caller from landline or mobile 
is hearing us very well, but the agent who is sitting on the handset is not 
hearing well, the voice at the agent is not crystal (like he is talking from 
well or far deep place). Although the IP Phones are cisco 7942G and the used 
codec is g711ulaw (actually it gave better quality than g711alaw).

If we increase the voice volume from the Cisco handset, the voice is becoming 
higher but more distortion, it is missing for the sharpness (clearance), we are 
hearing it like he is talking from well (from far and deep place).

I am trying to fix this .. I requested the provider to check if the problem 
from his telephony card, but it is not doing any thing.

What parameters can help me to fix this? trustrip, insecure, jitter, .. etc? 
Any of this can help?

Regards
Bilal

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[asterisk-users] Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger

[bilal ghayyad]

Hello;

Is there any modules or channels or integration between asterisk and any of the 
following:

whatsapp, facebook, viber, yahoo and hotmail messanger?

Regards
Bilal

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[asterisk-users] Cisco 7942G and SEPMAC.cnf.xml and the registration

[bilal ghayyad]

Hello;

I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The 
firmware has been downloaded from the TFTP successfully and currently I am 
running this load SIP42.9-3-1SR2-1S*

I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know 
which one to be used exactly. Basically, there is some effect that appears on 
the Phone (for example, it is appearing the extension on the button), but the 
Phone is not able to register. I tried to ssh or even http or https to the 
phone but I can not access it. Although I configured the ssh in the 
SEPMAC.cnf.xml as following:

sshUserIdadmin/sshUserId
sshPasswordcisco/sshPassword

Anyone tried to register Cisco 7942G on Asterisk? Which SEPMAC.cnf.xml was used?

How I can access the Phone via ssh or http to be able to see the logs and 
understand what is happening?

By the way: this SEPMAC.cnf.xml is existed on cisco website? Is it specialized 
for each Phone type (does it differs from Cisco IP Phone 7940 to 7942 to 7960)? 

Appreciate the help as really I am sticked at this point and not able to 
moveforward.

Thanks in advance.
Regards
Bilal

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Re: [asterisk-users] Cisco 7942G and SEPMAC.cnf.xml and the registration

[bilal ghayyad]

Hello;

The phones are registering now. I found a SEPMAC.cnf.xml file and I used sip 
firmware version 8.3 and I configured nat=no at sip.conf and nat to be false in 
xml file.

But I am facing a time problem, I am in Kuwait country and the time that is 
appearing at the Phones screen is delayed by 3 hours. Kuwait time is GMT+3.

Anyone can help?

Now I am placing the following in the xml file (but I am sure it needs to be 
corrected, how I do not know):

dateTimeSetting
dateTemplateD/M/Y/dateTemplate
timeZoneKuwait/timeZone
ntps
ntp
namepool.ntp.org/name
ntpModeUnicast/ntpMode
/ntp
/ntps
/dateTimeSetting



Regards
Bilal


 
 Hello;
 
 I am facing a problem to let Cisco IP Phone 7942G register
 on Asterisk. The firmware has been downloaded from the TFTP
 successfully and currently I am running this load
 SIP42.9-3-1SR2-1S*
 
 I feel that there is a problem in the SEPMAC.cnf.xml but
 really I do not know which one to be used exactly.
 Basically, there is some effect that appears on the Phone
 (for example, it is appearing the extension on the button),
 but the Phone is not able to register. I tried to ssh or
 even http or https to the phone but I can not access it.
 Although I configured the ssh in the SEPMAC.cnf.xml as
 following:
 
 sshUserIdadmin/sshUserId
 sshPasswordcisco/sshPassword
 
 Anyone tried to register Cisco 7942G on Asterisk? Which
 SEPMAC.cnf.xml was used?
 
 How I can access the Phone via ssh or http to be able to see
 the logs and understand what is happening?
 
 By the way: this SEPMAC.cnf.xml is existed on cisco website?
 Is it specialized for each Phone type (does it differs from
 Cisco IP Phone 7940 to 7942 to 7960)? 
 
 Appreciate the help as really I am sticked at this point and
 not able to moveforward.
 
 Thanks in advance.
 Regards
 Bilal

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Re: [asterisk-users] Sending SMS from asterisk

[bilal ghayyad]

Hi Asghar;

I was looking to use chan_mobile for sending SMS, is it possible? Or it is only 
for calls? 

By the way, if I have GSM adaptor that convert from SIM card to FXS port, then 
who I need chan_mobile? I can use DAHDI. So when to use chan_mobile? 

Regards
Bilal

-
 
 HI Bilal,
 i am using chan_mobile for call termination, you can use it
 but you need
 to tweak chan_mobile.c it is broken from a long time.
 let me know if you want give it a try.
 
 On Mon, Mar 11, 2013 at 6:22 PM, bilal ghayyad bilmar...@yahoo.com
 wrote:
 
  -
What are the elements of this solution? Is it
 only: 3G
   dongles and chan_dongle only? Or there are
 something else?
  
   Bash and perl programing, asterisk and
 chan_dongle.
  
 
  * Bash and perl programing to do what? It is going to
 use AMI instead of
  sending the messages from the commands given in the
 extensions.conf?
 
  Why to use chan_dongle and not chan_mobile?
 
  Regards
  Bilal

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Re: [asterisk-users] Sending SMS from asterisk

[bilal ghayyad]

-
  What are the elements of this solution? Is it only: 3G
 dongles and chan_dongle only? Or there are something else?
 
 Bash and perl programing, asterisk and chan_dongle.
 

* Bash and perl programing to do what? It is going to use AMI instead of 
sending the messages from the commands given in the extensions.conf?

Why to use chan_dongle and not chan_mobile?

Regards
Bilal

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Re: [asterisk-users] digium card and virualbox

[bilal ghayyad]

I am not mixing. I need this for LAB testing. 
How? This PCI passthrough, how to enable it on virualbox?
---
  Hi All;
 
  How to let the virualbox (ubuntu OS) to be able to see
 the digium card? Because when I install elastix or asterisk
 with dahdi, it is not able to see the digium card if the
 installation though the virualbox .. What is the solution?
 The solution is to run Ubuntu and Asterisk on your hardware
 natively, 
 not through VirtualBox.
 
 Virtualisation and high-performance hardware such as
 telephony cards  
 (it will be creating 8000 interrupts per phone line per
 second)  do not 
 mix, I am afraid.
 
 --
 AJS.

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Re: [asterisk-users] digium card and virualbox

[bilal ghayyad]

Yes I installed Elastix on virualbox but was not able to capture the digium 
card. But really I did not check well how is the overall situation (accessing 
from web based, if any IP Phone can register, .. etc).

If ubuntu can see the digium or not, also I did not try this. But meanwhile, I 
am using ubuntu 12.04 LTS. I do not know if someone who used Ubuntu can advise 
us in this issue?

Regards
Bilal

---
 
 hello
 
 regardless the virtual box, just in terms of Ubuntu, I have
 experience that
 Digium TP110p does now work with Ubuntu. it was long time a
 go I had this
 experience, I hardly could remember that what Ubuntu version
 I was using.
 my experience was on the Ubuntu system would not able to
 load DAHDI driver.
 for example:
 if you issue command
 dmesg |grep TE110
 then it would say
 wct1xxp :04:00.0: Not Found something..
 hope my experience would help you something
 
 
 by the way did you install Elastix in the  virtual box
 ?
 
 Sent from Shitian Long
 
 
 On Mar 8, 2013, at 10:21 PM, bilal ghayyad bilmar...@yahoo.com
 wrote:
 
 Hi All;
 
 How to let the virualbox (ubuntu OS) to be able to see the
 digium card?
 Because when I install elastix or asterisk with dahdi, it is
 not able to
 see the digium card if the installation though the virualbox
 .. What is the
 solution?
 
 Regards
 Bilal 

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Re: [asterisk-users] Sending SMS from asterisk

[bilal ghayyad]

Dears;

We have running here a SMS solution with four 3G dongles, which sends over 
20.000 SMS a month.

What are the elements of this solution? Is it only: 3G dongles and chan_dongle 
only? Or there are something else?

About the script that you wrot it: 

This script is using asterisk (through AMI) to send the SMS? Or it is working 
without need for asterisk? In this case, where is the benifit of using Asterisk 
to send the SMS?

Regarding to Sangoma and khomp: Do u mean that they have something like Huwewi 
3G dongles?

Regards
Bilal

--
 
 
 Hi Bilal,
 
 It's not necessary to use a FXS port, you can compile 
 install 
 chan_dongle and buy a Huawei 3G dongle.
 We have running here a SMS solution with four 3G dongles,
 which sends 
 over 20.000 SMS a month.
 
 In addition, I wrote an script able to send up to 12000
 characters in 
 concatenated SMS (the recipient receives a single SMS only)
 
 chan_dongle works very well.
 
 -- 
 ==
 Miguel Oyarzo
 Senior [ Network | Systems Design ] Engineer
 http://www.linkedin.com/in/mikeaustralia
 Linux User: # 483188 - counter.li.org
 Melbourne, Australia
 
 
 On 3/9/2013 1:09 PM, Gerardo Barajas wrote:
  Yes, you can check solutions from sangoma and khomp.
 
  Saludos/Regards
  --
  Ing. Gerardo Barajas Puente
 
  Proyectos Especiales/Preventa | www.neocenter.com 
  http://www.neocenter.com
  T:+52 (55)  8590-9000 x 7003
 
 
  On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com
 
  mailto:bilmar...@yahoo.com
 wrote:
 
      Hi;
 
      If my landline service provider
 does not provide the ability to
      send the SMS, and I need to
 send SMS from asterisk, then what is
      the required? How?
 
      Is it possible to send SMS from
 asterisk using SIM card to be
      connected via GSM adaptor
 connected to FXS ports? Or HOW?
 
      From the other side, this is
 existed only in asterisk 1.8 or it is
      existed in asterisk 1.4?
 
      Regards
      Bilal

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Re: [asterisk-users] digium card and virualbox

[bilal ghayyad]

Hello Gertjan;

I've heard a lot about it but I'm running Asterisk on ESXi5 Dell boxes without 
problems

* How your ESXi saw the digium? Is it using PCI Passthru?

Regards
Bilal

- 
  It's called PCI Passthru and from what I've tried, the
 timing is horrible in a virtualized environment. 
 VirtualBox and ESXi 5
 
  Doug
 
 What are your experiences, Doug. I've heard a lot about it
 but I'm
 running Asterisk on ESXi5 Dell boxes without problems. Did
 you
 encouter the timing issues with a lot of concurrent calls?
 Where the
 boxs slammed bij other vm's at the time?
 
 --Gertjan
 
 
 On Fri, Mar 8, 2013 at 10:29 PM, Doug Lytle supp...@drdos.info
 wrote:
  How to let the virualbox (ubuntu OS) to be able
 to see the digium card?
 
  It's called PCI Passthru and from what I've tried, the
 timing is horrible in a virtualized environment. 
 VirtualBox and ESXi 5
 
  Doug

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[asterisk-users] digium card and virualbox

[bilal ghayyad]

Hi All;

How to let the virualbox (ubuntu OS) to be able to see the digium card? Because 
when I install elastix or asterisk with dahdi, it is not able to see the digium 
card if the installation though the virualbox .. What is the solution?

Regards
Bilal

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[asterisk-users] Sending SMS from asterisk

[bilal ghayyad]

Hi;

If my landline service provider does not provide the ability to send the SMS, 
and I need to send SMS from asterisk, then what is the required? How?

Is it possible to send SMS from asterisk using SIM card to be connected via GSM 
adaptor connected to FXS ports? Or HOW?

From the other side, this is existed only in asterisk 1.8 or it is existed in 
asterisk 1.4?

Regards
Bilal

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[asterisk-users] Elastix vs vicidial

[bilal ghayyad]

Hi;

I used vicidial for call center and I would like to try elastix. Can someone 
advise about the advantages?

Does Elastix has a screen for the agent to login/logout from their PC and deal 
with the inbound/outbound calls and Integrated with the CRM?

Regards
Bilal

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[asterisk-users] RTP timeout if the asterisk box behind NAT

[bilal ghayyad]

Dears;

I am facing a problem in disconnecting the calls, it is related to the 
rtptimeout (disconnecting if there is no RTP packets from both sides).

My Asterisk Box is behind NAT but there is a static real IP address at the ADSL 
router. We call from the Mobile to the PSTN analogue numbers which are 
connected to Asterisk Analogue card (the telephone lines are analoge), and then 
we dial the overseas number, so the asterisk is sending the call to a VoIP 
service provider which will route the call to the destination. Sometime the 
destination is connected while ringing !! And this is a problem from the SIP 
service provider route, then we hangup our mobile (as no one answering our 
call) but asterisk is not detecting the hangup (it is because the telephone 
lines are analoge and this problem is common in analoge lines that some hangup 
are not detected). In that case, the call will stay open and charging and this 
is a wrong.

This problem was not appearing when Asterisk machine was having static real IP 
address because I was enabling the rtptimeout paramters. But now as the 
asterisk box IP address is private and it is behind NATing then it is appearing 
even I enabled the (rtptimeout=50 and rtpholdtimeout=120).

What should I do?

Regards
Bilal

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[asterisk-users] How to assign the button on the IP Phone to a feature?

[bilal ghayyad]

Dear;

Using Cisco IP Phones: How I can assign a button for a function. For example, 
if we pressed on this button, then we need to pickup the call from the group.

Another thing:

If the button pressed, then the call forward function to be enabled (and it 
should appear on the phone that it is grayed which indicates that it is 
enabled). So, when pressing this button, it means we are making the function ON 
and it is appearing at the LCD to be gray. And if we did another press on the 
button, then the gray to be removed and that means the function is OFF (so no 
call forward will happen). HOW?


Generally speaking: How I can play in cisco button to assign them for features 
and controlling their appearance?

Regards
Bilal

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Re: [asterisk-users] Integration with Social Media, Email and Web call center

[bilal ghayyad]

They advised me to check jabber.org.
Yes, jabber.org has a client that can send/receive and integrate with other 
social media (facebook, msn, twitter, ... etc).

But, as an Agent who can login/logout and take a calls, how can I make it to be 
single login for voice and messages. So, if the agent is not available, he will 
not get a calls and will not get a messages.

Those who used jabber.org or who used other than jabber.org for such 
requirement, what do you suggest? 

Regards
Bilal

--

 
 For just the messaging part, you should be able to use wget
 or curl to
 interface and create messages.  You might have to go a
 little higher level
 like C or Perl, but it sounds very doable.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of bilal ghayyad
 Sent: Tuesday, January 22, 2013 4:27 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Integration with Social Media,
 Email and Web call
 center
 
 Dears;
 
 Can someone advise me where to find a technology (open
 source) that let us
 able to integrate with social media like whatsapp and
 facebook? And use this
 in call center (queuing the messages and routing it for
 agent)?
 
 Anyone give me a light to start?
 
 Regards
 Bilal

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Re: [asterisk-users] How to assign the button on the IP Phone

[bilal ghayyad]

Both: SPA and 7900. let us say 7942. How?

Regards
Bilal


 
  Dear;
 
  Using Cisco IP Phones: How I can assign a button for a
 function. For
  example, if we pressed on this button, then we need to
 pickup the call from
  the group.
 
 
 Which model line?  The SPA series, or the 7900 and
 similar?  They are
 completely different.

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[asterisk-users] Integration with Social Media, Email and Web call center

[bilal ghayyad]

Dears;

Can someone advise me where to find a technology (open source) that let us able 
to integrate with social media like whatsapp and facebook? And use this in call 
center (queuing the messages and routing it for agent)?

Anyone give me a light to start?

Regards
Bilal 



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[asterisk-users] Email and web chat call center

[bilal ghayyad]

Dears;

I am using asterisk for call center and I used also VICIDIAL. But they are fine 
for voice, I need the agents to be able to handle email and web chat messages 
as long with the voice calls, in addition to be integrated with the social 
media like Facebook and twitter.

Where I can find this? From where I can read and start? Is there a reliable 
open source solution for this?

Regards
Bilal--
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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Thanks for the help and it seems I deleted some of my emails by mistake ! I am 
sorry if I repeated my question.

As I see that the call file is used to generate calls, can I use this technique 
to page the Phones?

It is one wave file only that need to be Paged for all the Phones connected on 
the Asterisk PBX.

When I say Paging, I mean that they are going to hear the sound from the 
speaker (without pickup the handset).

By using AMI, then I can build PHP script that will use the AMI to do the Page?

Thanks and Regards
Bilal

 --
 

  A call file is a text file that you create. The
 format is very 
  specific.
 
 On Tue, 1 Jan 2013, bilal ghayyad wrote:
 
  * How can I know this format? Because I need to know
 what should I place 
  in this file so it will execute Paging for this group
 of Phones?
 
 This may help:
 
      http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
 
  How many customers will be receiving these
 reminders?
 
  * It is required that all the employers at the company
 to hear this on 
  their IP Phones.
 
 In my experience, you can't just dump xxx call files into
 the outgoing 
 directory. If you expect more than a dozen or so, you'll
 have to move them 
 in blocks as they are processed. Another good reason to use
 AMI.
 
  You can 'schedule' a call file to be processed in
 the future by setting 
  the file's 'mtime.'
 
  * Can you explain for me please?
 
 Create a file named fajr containing:
 
      application:    playback
      channel:    sip/bilal
      data:   
     fajr-in-10-minutes
 
 Copy the file to a directory we assume is on the same file
 system as 
 /var/spool/asterisk/outgoing/:
 
      cp\
          fajr\
         
 /var/spool/asterisk/tmp/
 
 Set the file's 'mtime'
 
      touch\
          --date='now + 2
 minutes'\
          --time=mtime\
             
 /var/spool/asterisk/tmp/fajr
 
 Move it to the outgoing directory:
 
      mv\
         
 /var/spool/asterisk/tmp/fajr\
         
 /var/spool/asterisk/outgoing/
 
 Your phone should ring in about 2 minutes.
 
 You may want to look into setting 'auto-answer' or some sort
 of 'overhead 
 paging' with a very discreet sound file like a short, single
 beep.
 
 Please consider AMI if you are looking for a robust
 service.
 
 -- 
 Thanks in advance,
 -
 Steve Edwards       sedwa...@sedwards.com 
     Voice: +1-760-468-3867 PST
 Newline             


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[asterisk-users] DAHDI: How to know since when it is used? How to shutdown after max time?

[bilal ghayyad]

Hi;

How can I know the duration that the DAHDI channel is still used? I need to 
know its status and since when it is in this status, how?

Also, is it possible to hangup the channel if it has been openned more than 90 
minute? Other than using the timeout in the Dial command (because this I know 
it).

What is happening with me that from time to time, I find some DAHDI channels 
are stayed connected (stuck) for long time. I know how to write the 
extensions.conf in a way to handle the hangup properly, also I send the 
incoming calls to the voicemail to be sure it is hanged up properly. One more 
thing, I set the rtptimeout in case there is any problem in the sip phone and 
the network .. But, still after sometime, I am surprised that some channels are 
stuck and stayed connected and then I have to reset it manually !! This is 
happening only in the analoge channels.

What other than the rtptimeout, the hangup in the extensions.conf, the 
voicemail? Is there anything I have to take care for it that might cause this 
stuck and keeping the channel openned?

By the way, for such cases, what should I place the value of the rtpkeepalive 
as currently it is 0?

What other things I have to take care for it?

Regards
Bilal

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Thanks for the help.

As I see that the call file is used to generate calls, can I use this technique 
to page the Phones?

It is one wave file only that need to be Paged for all the Phones connected on 
the Asterisk PBX.

When I say Paging, I mean that they are going to hear the sound from the 
speaker (without pickup the handset).

By using AMI, then I can build PHP script that will use the AMI to do the Page?

Thanks and Regards
Bilal

 
  A call file is a text file that you create. The
 format is very 
  specific.
 
 On Tue, 1 Jan 2013, bilal ghayyad wrote:
 
  * How can I know this format? Because I need to know
 what should I place 
  in this file so it will execute Paging for this group
 of Phones?
 
 This may help:
 
      http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
 
  How many customers will be receiving these
 reminders?
 
  * It is required that all the employers at the company
 to hear this on 
  their IP Phones.
 
 In my experience, you can't just dump xxx call files into
 the outgoing 
 directory. If you expect more than a dozen or so, you'll
 have to move them 
 in blocks as they are processed. Another good reason to use
 AMI.
 
  You can 'schedule' a call file to be processed in
 the future by setting 
  the file's 'mtime.'
 
  * Can you explain for me please?
 
 Create a file named fajr containing:
 
      application:    playback
      channel:    sip/bilal
      data:   
     fajr-in-10-minutes
 
 Copy the file to a directory we assume is on the same file
 system as 
 /var/spool/asterisk/outgoing/:
 
      cp\
          fajr\
         
 /var/spool/asterisk/tmp/
 
 Set the file's 'mtime'
 
      touch\
          --date='now + 2
 minutes'\
          --time=mtime\
             
 /var/spool/asterisk/tmp/fajr
 
 Move it to the outgoing directory:
 
      mv\
         
 /var/spool/asterisk/tmp/fajr\
         
 /var/spool/asterisk/outgoing/
 
 Your phone should ring in about 2 minutes.
 
 You may want to look into setting 'auto-answer' or some sort
 of 'overhead 
 paging' with a very discreet sound file like a short, single
 beep.
 
 Please consider AMI if you are looking for a robust
 service.

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

 How many customers will be receiving these reminders?

* It is required that all the employers at the company to hear this on their IP 
Phones.

 What religion is this targeted to?

* Islam.


 A call file is a text file that you create. The format is
 very specific. 

* How can I know this format? Because I need to know what should I place in 
this file so it will execute Paging for this group of Phones?


 You can 
 'schedule' a call file to be processed in the future by
 setting the file's 
 'mtime.'

*  Can you explain for me please?

I am fully thanks. 

Regards
Bilal

--

 
  I have one more question:
 
  What was u meaning by call file and why it is required
 to place them in 
  the 'astspooldir.'?
 
 There are 2 methods of originating a call external to
 Asterisk: call files 
 and the Asterisk Manager Interface (AMI).
 
 A call file is a text file that you create. The format is
 very specific. 
 It could contain (in the context of your needs) the phone
 number to dial 
 and the path of the file to play. A call file is kind of
 like a 'message 
 in a bottle.' You cast it into the sea and hope for the
 best. When this 
 file is mv'ed into the directory specified in the Asterisk
 astspooldir 
 variable, Asterisk will read it and try to do what you want.
 You can 
 'schedule' a call file to be processed in the future by
 setting the file's 
 'mtime.'
 
 The Asterisk Manager Interface (AMI) is a TCP connection
 between your 
 program and Asterisk. You can issue commands (like
 originate) and receive 
 responses. AMI is more robust because you can make decisions
 based on the 
 response.
 
 If robustness is not of primary importance, a script
 scheduled by cron to 
 run after midnight could create the 5 call files for that
 day, setting the 
 'mtime' of each file before mv'ing the file to the
 directory specified 
 by astspooldir -- usually /var/spool/asterisk/outgoing/
 
 How many customers will be receiving these reminders?
 
 What religion is this targeted to?

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Dear Steve;

Thanks a lot for your help.

Specifically what I need is the following:

1) One wave file to be paged at the Phones. In the 5 times, the same file will 
be used.

2) Praying time need to be obtained from text (or database). So, it is not 
always the same time. What actually is needed to be obtained from the text file 
or the database is the time of the pray for each date (for example, if today is 
28 of December so the query will be for this date and then it is required to 
check if the time is same as the current time to page the wave file on the 
Phones).

I have one more question:

What was u meaning by call file and why it is required to place them in the 
'astspooldir.'?

Regards
Bilal

 Please don't top-post.
 
 On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad bilmar...@yahoo.com
 
 wrote:
 
 How can I have Paging on Asterisk to call for pray?
 
 The pray is 5 times in a day and there is a timing for pray
 (actually it 
 can be existed in a text file or database for the next 2 or
 5 years).
 
 [mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Bharat 
 Lalcheta
 
 However, its easy to build a script in php or perl or any
 other language 
 which check time from file or database and generate call
 file which 
 execute paging in asterisk. Just put this script in cron.
 Thats it...
 
 From: Danny Nicholas da...@debsinc.com
 
 I would set up 5 shell files called pray1.sh, pray2.sh, etc
 and then set 
 up 5 entries in /etc/crontab to run them at the specified
 time daily. The 
 file pray1.sh should look something like this:
 
 #!/bin/sh
 cp /pray1/*.call /tmp
 mv /tmp/*.call /var/spool/asterisk/outgoing
 
 the entry in /etc/crontab would look like this
 
 0 8 *** root /usr/bin/pray1.sh
 
 This would run pray1.sh at 8 am daily.
 
 On Thu, 27 Dec 2012, bilal ghayyad wrote:
 
  Thanks a lot for your kindly reply and help.
 
  Really I did not understand why you need to place them
 in the 
  /var/spool/asterisk/outgoing?
 
 The appropriate solution needs a lot more detail to be
 useful.
 
 Is this just to remind you or is this the foundation of a
 new product for 
 thousands of customers?
 
 Is there a message or verse associated with each of the 5
 reminders or is 
 'time to pray' sufficient?
 
 Is there a penalty associated with missing a prayer like
 eternal 
 damnation? (AMI is more robust than call files.)
 
 The answers would help guide you in deciding if a simple
 cron based shell 
 script generating call files or a database driven AMI daemon
 is the best 
 approach.
 
 In answer to your specific question, the call files need to
 be mv'ed
 into /v/s/a/o/ because:
 
 ) You need to use mv instead of cp because mv is an 'atomic'
 function* 
 meaning it happens all at once so that Asterisk will not try
 to read an 
 incomplete file.
 
 ) This is the default value of 'astspooldir.' You can
 specify a different 
 location in asterisk.conf if needed.
 
 *) Assuming the source and destination are on the same
 filesystem.

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Re: [asterisk-users] Paging for Praying

[bilal ghayyad]

Thanks a lot for your kindly reply and help.

Really I did not understand why you need to place them in the 
/var/spool/asterisk/outgoing?

Regards
Bilal


---
 
 I would set up 5 shell files called pray1.sh, pray2.sh, etc
 and then set up
 5 entries in /etc/crontab to run them at the specified time
 daily.  The file
 pray1.sh should look something like this:
 
 #!/bin/sh
 
 cp /pray1/*.call /tmp
 
 mv /tmp/*.call /var/spool/asterisk/outgoing
 
  
 
 the entry in /etc/crontab would look like this
 
 0 8 *** root /usr/bin/pray1.sh
 
  
 
 This would run pray1.sh at 8 am daily.
 
  
 
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Bharat
 Lalcheta
 Sent: Thursday, December 27, 2012 2:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Paging for Praying
 
  
 
 I dont think this is existed.
 
  
 
 However, its easy to build a script in php or perl or any
 other language
 which check time from file or database and generate call
 file which execute
 paging in asterisk. Just put this script in cron. Thats
 it...
 
  
 
 Regards,
 
  
 
 Bharat Lalcheta
 
  
 
 
 
  
 
 On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad bilmar...@yahoo.com
 wrote:
 
 Hello;
 
 How can I have Paging on Asterisk to call for pray?
 
 The pray is 5 times in a day and there is a timing for pray
 (actually it can
 be existed in a text file or database for the next 2 or 5
 years).
 
 My question is compound from two parts:
 
 How can I have Automatic Page?
 
 The automatic page should happens by reading the time and
 check if the time
 is same as this time, then do the Page. How? Is it by cron?
 
 Someone told me that do a cron that call a script which will
 check the time,
 if the time came to do th Page, then do a Page. But really I
 do not know how
 this can be done and I do not know if this is already
 existed?
 
 Regards
 Bilal


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[asterisk-users] Paging for Praying

[bilal ghayyad]

Hello;

How can I have Paging on Asterisk to call for pray?

The pray is 5 times in a day and there is a timing for pray (actually it can be 
existed in a text file or database for the next 2 or 5 years).

My question is compound from two parts:

How can I have Automatic Page?

The automatic page should happens by reading the time and check if the time is 
same as this time, then do the Page. How? Is it by cron?

Someone told me that do a cron that call a script which will check the time, if 
the time came to do th Page, then do a Page. But really I do not know how this 
can be done and I do not know if this is already existed?

Regards
Bilal

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[asterisk-users] Calling from SIP client then bridge between two end points

[bilal ghayyad]

Hi All;

How I can acheive the following:

From sip client softphone (from the iPhone for example), if I dialed a number 
that I need to call it, then a call to be initated to a specific number 
through DAHDI channel and another call to be initiated for the destination 
number (the number that I dialed it from the softphone) and these two calls to 
be linked togethor (to call each other directly). So the call from the 
softphone just to important in the begining to trigger this scenario.

How this settings to be done?

Regards
Bilal

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[asterisk-users] SDK for Asterisk, where is it?

[bilal ghayyad]

Hello;

I remember that I saw at asterisk website (this was maybe before 1 year or 
around) some pages are talking about having SDK and APIs for asterisk that will 
be used to build softphone for mobile and will be used to build some 
applications for asterisk, also it was mentioned in this page that this project 
is resuming after it was stopped before. But now I am not able to find those 
pages about this subject any more. What happened about it? Are they going to 
really do it or it is cancelled.

If it is still existed, where is the link for this and what it is situation?

Regards
Bilal

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[asterisk-users] Allowing peers from specific subnet only

[bilal ghayyad]

Hi;

How I can make my configuration to allow the sip phones only from specific IP 
addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed 
to connect for asterisk?

In other words, in addition to be authenticated based on the username and 
password, it is required that the IP address of the Phone to be from this 
range. How?

Regards
Bilal

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[asterisk-users] Sending calls from behind NAT

[bilal ghayyad]

Dears;

It seems my service provider is requesting a complicated settings to allow me 
to send from behind NAT. 

What they said:

It shouldn't matter as long as you are handling the NAT correctly your end. We 
do not fix NAT so if you're sending internal addresses in your INVITEs or SDP 
then things will fail but if you're handling it correctly, we shouldn't tell 
the difference.


Really, I did not understand what exactly they need. But maybe what they need 
is to see my public IP address without the private IP address (this what I 
understood if I am right).

I tried to use the following in the [general] settings in the sip.conf

localnet=192.168.10.2/255.255.255.254
externadd =196.40.164.239

But even, the calls are drop .. so what I have to do?

The following what I get when I enabled the sip debug:


--- SIP read from UDP:194.0.220.220:5060 ---
SIP/2.0 403 UA behind NAT not accepted here
Via: SIP/2.0/UDP 
192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239
From: asterisk sip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b
To: 
sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5
Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060
CSeq: 102 INVITE
P-Behind-NAT: source
Server: Service Provider Global Proxy v2
Content-Length: 0

So what could resolve my problem?

Regards
Bilal

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Re: [asterisk-users] Sending calls from behind NAT

[bilal ghayyad]

Dears;

What Jian said is the right and it worked.

But I have the following questions:

Why 192.168.10.2 is wrong and I have to use 192.168.10.0? Also, do I have to 
set the localnet or it is enough to set the externip?

From the other side, I am using Asterisk 1.8.12.0 and when I was searching in 
the sip.conf, I did not find externip (so I added by my hand) and I remember 
very well that before I was able to find the externip in the sip.conf, 
although I am finding externadd. So why this? 

One more thing, what is the difference between externadd and externip?

Regards
Bilal

---
  Dears;
 
  It seems my service provider is requesting a
 complicated settings to allow me to send from behind NAT.
 
  What they said:
 
  It shouldn't matter as long as you are handling the
 NAT correctly your end. We do not fix NAT so if you're
 sending internal addresses in your INVITEs or SDP then
 things will fail but if you're handling it correctly, we
 shouldn't tell the difference.
 
 
  Really, I did not understand what exactly they need.
 But maybe what they need is to see my public IP address
 without the private IP address (this what I understood if I
 am right).
 
  I tried to use the following in the [general] settings
 in the sip.conf
 
  localnet=192.168.10.2/255.255.255.254
  externadd =196.40.164.239
 
 
 
 I think these setting are all wrong:
 1. local network should be something like: 192.168.10.0
 2. Subnetmask cant' be 255.255.255.254 !
 3. externip=x.x.x.x (Not externadd)
 
 Jian
 
  But even, the calls are drop .. so what I have to do?
 
  The following what I get when I enabled the sip debug:
 
 
  --- SIP read from UDP:194.0.220.220:5060 ---
  SIP/2.0 403 UA behind NAT not accepted here
  Via: SIP/2.0/UDP
 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239
  From: asterisk
 sip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b
  To: 
  sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5
  Call-ID:
 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060
  CSeq: 102 INVITE
  P-Behind-NAT: source
  Server: Service Provider Global Proxy v2
  Content-Length: 0
 
  So what could resolve my problem?
 
  Regards
  Bilal


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[asterisk-users] Automatic jump from line to line for incoming calls and the problem in DAHDI

[bilal ghayyad]

Dears;

I am facing the following problem:

Already we requested from the service provider to enable the auto jumping 
service for our analoge telephone lines, so because we have 4 telephone lines 
from the service provider, then if you called line # 1 and it was busy, then 
the call will be sent to any available line #2 or #3 or #4, and if you call 
line # 3 and it was busy then the call will be sent via any available line of 
these four lines.

This feature is causing a problem at the Asterisk PBX, so some calls are not 
handled properly (it is ringing and we do not hear the welcome message), also 
the outgoing calls are facing a problem because it seems that there is a 
confusing happening in dahdi to determine the available line.

I do not know really how the automatic jumping feature is working at the 
service provider and what is the effecting on the DAHDI and Asterisk that is 
causing to not responding for the DAHDI channels properly.

For more details to be sure that I described the behaviour of the auto jumping 
feature that I took it from the service provider, let us assume my number is 
22446789, when I call this number and I look for asterisk CLI, I can see that 
the call came via DAHDI/3-1 and then I do another call to this line, I can see 
it via DAHDI/4-1 and I do another call to this line and I will see it via 
DAHDI/2-1.

Also, not all my calls are failed ... but some are succeed and some are fails, 
so the responding is not perfect. I am sure because of the auto jumping feature 
from the service provider. 

Appreciate the kindly help and advise.

Regards
Bilal

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Re: [asterisk-users] Automatic jump from line to line for incoming calls and the problem in DAHDI

[bilal ghayyad]

Actually I am not talking on how to handle it in the extensions.conf because I 
am doing same as you wrote. But even so, I am facing a problem that some calls 
are captured and some calls are not captured.

Currently, I set the callwaiting=no in the chan_dahdi.conf, it seems it is 
working fine. But I am not sure if this is really the required configuration to 
fix it or there is something else.

Any advise.

Regards
Bilal


  Dears;
  
  I am facing the following problem:
  
  Already we requested from the service provider to
 enable the auto jumping
  service for our analoge telephone lines, so because we
 have 4 telephone
  lines from the service provider, then if you called
 line # 1 and it was
  busy, then the call will be sent to any available line
 #2 or #3 or #4, and
  if you call line # 3 and it was busy then the call will
 be sent via any
  available line of these four lines.
  
  This feature is causing a problem at the Asterisk PBX,
 so some calls are
  not handled properly (it is ringing and we do not hear
 the welcome
  message), also the outgoing calls are facing a problem
 because it seems
  that there is a confusing happening in dahdi to
 determine the available
  line.
  
  I do not know really how the automatic jumping feature
 is working at the
  service provider and what is the effecting on the DAHDI
 and Asterisk that
  is causing to not responding for the DAHDI channels
 properly.
  
  For more details to be sure that I described the
 behaviour of the auto
  jumping feature that I took it from the service
 provider, let us assume my
  number is 22446789, when I call this number and I look
 for asterisk CLI, I
  can see that the call came via DAHDI/3-1 and then I do
 another call to
  this line, I can see it via DAHDI/4-1 and I do another
 call to this line
  and I will see it via DAHDI/2-1.
  
  Also, not all my calls are failed ... but some are
 succeed and some are
  fails, so the responding is not perfect. I am sure
 because of the auto
  jumping feature from the service provider.
 
 If you have multiple lines, and they are all paid for in the
 same name, then 
 your telco really should have set it up so they are all
 accessible by dialling 
 the same number.
 
 Way back in the clicky-clicky days, having multiple lines
 connected to the 
 same switchboard would have been done at the exchange by
 allocating sequential 
 lines on the same selector, which was modified to step on
 until it found a non-
 engaged line  (or go to engaged tone, if the last in
 the set were engaged).  
 For instance, Radio Derby's main switchboard number was
 36; but 361112, 
 361113, 361114, 361115 or 361116 might also reach the
 switchboard  (depending 
 whether or not that line was already in use).
 
 Digital exchanges don't have such requirements, of course;
 and since we went 
 over to System X, which does not impose a 1:1 mapping
 between  (logical)  
 numbers and  (physical)  lines, 361112  (at
 least)  has been allocated to 
 another subscriber.  And there are many lines numbered
 36.
 
 If you have several lines and they are properly grouped by
 the telco, you may 
 get a call coming in via a differently-numbered line than
 what the other 
 subscriber actually dialled.  
 
 The way top deal with this in Asterisk is as follows: 
 Have one context that 
 handles incoming calls from the PSTN  (usually 
 [from-pstn]  but you may have 
 changed this).  In this context, you just need to
 handle calls for any 
 extension the same.  (Or make sure, by using a
 catch-all such as the s 
 extension or _X.)
 
 For calling out, make sure all your DAHDI channels are in
 the same group in 
 chan_dahdi.conf, and use something in your Dial() command
 like 
 Dial(DAHDI/g1/${EXTEN}) or Dial(DAHDI/r1/${EXTEN}) . 
 The g form will try 
 always to use the lowest-numbered available channel; the r
 form will keep a 
 track of which channel was used last and try to cycle
 through channels in turn 
 from lowest to highest.  (Capital G1 and R1 will try
 always to use the highest 
 number, and cycle through from high to low respectively).
 
 
 -- 
 AJS
 
 Answers come *after* questions.

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Re: [asterisk-users] SRTP asterisk 1.8.x SNOM

[bilal ghayyad]

Dear AJS;

I have fedora core 14 and I did yum install libsrtp-devel and it is already 
existed, the only thing happened that it updated it.

Currently:
libsrtp-1.4.4-1.20101004cvs.fc14.i686
libsrtp-devel-1.4.4-1.20101004cvs.fc14.i686

Again, I did make menuselect and the same problem, I am not able to select the 
SRTP to include it in the compilation.

Any advise to be able to have this module?


Regards
Bilal


-

  Hi;
  
  It seems the SNOM Phones are requesting to have SRTP
 but I do not have the
  module res_srtp.
  
  I tried to compile it with asterisk 1.8, make
 menuselect, but I found that
  it can not be used (I am not able to select it) with
 the following
  details:
  
  Secure RTP SRTP
  Depends on: srtp E
  Can use: N/A
  Conflicts with: N/A
  
  So, how I can use it?
  What I have to do to know the reason for not being able
 to use it?
 
 Things not compiling despite their dependencies apparently
 being satisfied is 
 almost always a sign of a missing developers' package 
 (and since you're 
 building software from source, you're considered a
 developer).  If you're 
 using Debian or Ubuntu, try
 
 $ sudo apt-get install libsrtp0-dev
 
 If you're using some RPM-based distro, they typically use
 -devel as a suffix to 
 indicate developers' packages.  Check your repository.
 
 -- 
 AJS
 
 Answers come *after* questions.

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[asterisk-users] SRTP asterisk 1.8.x SNOM

[bilal ghayyad]

Hi;

It seems the SNOM Phones are requesting to have SRTP but I do not have the 
module res_srtp.

I tried to compile it with asterisk 1.8, make menuselect, but I found that it 
can not be used (I am not able to select it) with the following details:

Secure RTP SRTP
Depends on: srtp E
Can use: N/A
Conflicts with: N/A

So, how I can use it? 
What I have to do to know the reason for not being able to use it?

From the other side, if I succeeded to compile the res_srtp and having it as 
supported channel on asterisk 1.8, the question is: it will be supported for 
SNOM or there will be a conflict?

Regards
Bilal

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[asterisk-users] Fax and sending to mail

[bilal ghayyad]

Hi All;

Is there a module (addon or already built in) that enable us to receive the fax 
on the telephony card and save it as image (or any other format) and sent it to 
email? 

Regards
Bilal

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Re: [asterisk-users] Showing the name of the called number at the source IP Phone, how?

[bilal ghayyad]

The IP Phones that I am using is the Digium D40, but did not find any place to 
enable the RPID, I do not know if they are enabled by default.

Any advise?
Regards
Bilal

--
 
 Hi,
 
 You need to set rpid on the calling phone settings, if that
 phone knows
 what to do with RPID. Then you need to set allowrpid=yes in
 the sip peer
 settings of A party and B party. I did that on CISCO 79X0
 phones and it
 worked perfectly,
 
 Regards,
 Sammy
 
 
 On Tue, Aug 7, 2012 at 3:43 AM, bilal ghayyad bilmar...@yahoo.com
 wrote:
 
  Hi All;
 
  Asterisk 1.8.11-cert1
 
  I need to do the following, how?
 
  If my extension is 500 and I need to call the extension
 501, so when
  dialing 501, then I need to be able to see the name of
 the 501 (for
  example, the name was: Mike, so I need to see at my IP
 Phone that I am
  calling Mike which is the name of the destination).
 
  How?
 
  Regards
  Bilal


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Re: [asterisk-users] Background, Playback wave files in asterisk

[bilal ghayyad]

Dears;

I discover that I have to place the wave files in the 
/var/lib/asterisk/sounds/custom/

So, can I understand that the only solution I have is to copy the files that 
are existed in the path /var/lib/asterisk/sounds/en/ to the path 
/var/lib/asterisk/sounds/custom? Or there is any other solution?

I am using FreePBX and the asterisk version is: Asterisk 1.8.11-cert1

Any advise?

Regards
Bilal

-
 
 Hello;
 
 What is the difference between using the Background 
 Playback in Asterisk 1.8 without cert and Asterisk 1.8
 cert?
 
 I surprised that in cert version, I do not hear the sound !
 And it is not working properly, but in the normal version,
 it is working.
 
 So what is the new?
 Is it the version? Or there are some variables or settings
 need to be done in asterisk 1.8 cert that was not require in
 the normal version (not cert)?
 
 Regards
 Bilal


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[asterisk-users] Showing the name of the called number at the source IP Phone, how?

[bilal ghayyad]

Hi All;

Asterisk 1.8.11-cert1

I need to do the following, how?

If my extension is 500 and I need to call the extension 501, so when dialing 
501, then I need to be able to see the name of the 501 (for example, the name 
was: Mike, so I need to see at my IP Phone that I am calling Mike which is the 
name of the destination).

How?

Regards
Bilal

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[asterisk-users] Background, Playback wave files in Asterisk 1.8.11-cert1

[bilal ghayyad]

Hello;

What is the difference between using the Background  Playback in Asterisk 1.8 
without cert and Asterisk 1.8 cert?

I surprised that in cert version, I do not hear the sound ! And it is not 
working properly, but in the normal version, it is working.

So what is the new?
Is it the version? Or there are some variables or settings need to be done in 
asterisk 1.8 cert that was not require in the normal version (not cert)?

Regards
Bilal

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[asterisk-users] Digium IP Phone D40 quality, very bad

[bilal ghayyad]

Hi All;

Really it is miserable.

I bring 8 Digium Phone D40 and I used them with a customer, the voice quality 
is bad internally (between the extension), there is no clearance at all ! We 
are hearing the voice like another person.

The used codec is ulaw.

The firmware version is: 1_1_0_0_48178

Even at the web based configuration at the phone it self, I am not able to do 
reboot (there is no reboot button) and I can do this only from the Phone it 
self.

From the speaker, the voice is very bad and weak.

I am really feel disappointed why I did not use Polycom.

Can someone help me or advise me what to do in this?
Regards
Bilal

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[asterisk-users] Asterisk and IPTV

[bilal ghayyad]

Dears;

Is it possible with Asteisk to have IPTV (ability to show the TV channels using 
the video over IP, but to be live). In other words, to use Asterisk to watch 
the TV Channels.

Which open source that can do this, so we can install it on the same asterisk 
machine?

Also, is it possible to use Asterisk as Video Service Provider, to boradcast 
the video channels for the clients?

Regards
Bilal

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Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

[bilal ghayyad]

Dears;

First, I would like to declare that I used sip_custom.conf and 
extensions_custom.conf and I discovered that the calls are not shown in the 
CDR. 

Until now I did not check if I can browse the voicemails using the FreePBX if I 
used the extensions_custom.com file.

Now, if we are talking about using the addon modules (official or not 
official), then it means still I have to stick on using the FreePBX GUI which 
generates complicated script.

As I see, if I need to have the CDR and the voicemail functionalities that can 
be browsed via the GUI, then I have to use the FreePBX or the third party 
modules and can not write manual in my hand as normally we do in the native 
Asterisk.

Thanks for the input for all the friends who shared with me and gave me the 
good information that also helped.

Regards
Bilal

 
 See
 Route-Permissions module,
 It lets you restrict certain phones/extensions to follow a
 dial-plan
 pattern and dial out to the defined trunk etc meanwhile not
 breaking any
 other functionality or features of FPBX- though you can
 restrict the
 features from this too.
 
 http://www.freepbx.org/support/documentation/howtos/how-to-give-a-particular-extension-different-or-restricted-trunk-access
 
 http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/outbound-route-permission
 
 http://mirror.freepbx.org/modules/release/contributed_modules/
 
 OR
 Custom Context
 http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/customcontexts
 
 
 See w/e fits your requirements. What I suggest suits your
 need is the
 Route-permission module. Though it'll be bit complicated but
 worth giving a
 try.
 
 Regards,
 Sammy
 
 
 On Thu, Jul 12, 2012 at 4:01 AM, Warren Selby wcse...@selbytech.com
 wrote:
 
  On Wed, Jul 11, 2012 at 4:56 PM, bilal ghayyad bilmar...@yahoo.comwrote:
 
  Fine, did you read the question well and understand
 about what I am
  asking?
 
 
  Perhaps I did not understand what you were
 asking.  I thought you were
  wanting to do something custom per extension (in the
 case of my example,
  the something custom was control outbound call access
 to either local
  only or local and long distance, etc.  You can
 figure out you're own
  something custom), but still have all the calls have
 all the standard
  FreePBX features that you only get when using the
 [from-internal] context.
 
  In my example, the extensions are in the 2XXX range,
 and they would either
  have a context of [custom-local-only] or
 [custom-long-distance], depending
  on what you wanted to allow that extension to dial.
 
  To break down my example:
 
 
 
  [custom-local-only]  -- The name of our custom
 context.  It could be
  anything you want, as long as it's in square brackets
 
  exten = _281NXX,1,Verbose(Outbound call from
 local-only context) --
  This step is purely informational, it has no bearing on
 CDRs or anything
  else...it's just a useful step for debugging.  I
 tend to do this for
  everything, it's the same as some people use the
 NoOp() command to have
  debugging information in their CLI output.
 
   same = n,Goto(${EXTEN},from-internal,1) 
 -- This step sends the call to
  the [from-internal] context and handles it exactly as
 if you weren't using
  any custom call controls.  In my example, however,
 it will only go there if
  it meets the criteria of matching the pattern (in other
 words, the call
  would have to be placed to a number that matches the
 _281NXX pattern).
  same = n is a shorthand way of writing exten
 = _281NXX,n.  It was
  added in around 1.6 I think, I'm not entirely sure.
 
  exten = _2XXX,1,Verbose(Internal
 extension-to-extension call)  -- Again,
  this is purely an informational step, useful for
 debugging.  It can be
  skipped or expanded as you see fit, it has no bearing
 on CDR records or
  anything else, other than CLI output.
 
   same = n,Goto(${EXTEN},from-internal,1) 
 -- This does the same as the
  previous example, however it will only go to the
 [from-internal] context if
  the pattern that was dialed matches _2XXX.  This
 is assuming you're using
  internal extensions in the range of _2XXX.  You
 can change this to whatever
  works for you.
 
  [custom-long-distance]  -- another custom
 context, this time it allows
  long distance NANPA calling as well as local and
 internal calls
 
  exten = _1NXXNXX,1,Verbose(Outbound call from
 local and long-distance
  context)  -- I hope you're seeing the pattern
 by now.  This is simply a
  useful debugging step, with no bearing on anything
 else.
 
   same = n,Goto(${EXTEN},from-internal,1) 
 -- The call passes into the
  [from-internal context if it matches the pattern of
 _1NXXNXX, a typical
  NANPA long distance call.
 
  include = custom-local-only  -- include
 the local dialing context that
  way we don't have to duplicate any code that we've
 previously written,
  mostly useful for the internal