How To Play Different Different Hold Music.

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Dear All, I have two server 'A' and 'B' . In Server 'A', five different ivr (Sevices) is playing and call is *forwarding *into Server 'B'. Server 'B' basically use for agent login(Extension). I want to play different hold music(Server 'B') bases on the corresponding services which is running into server 'A'. A single agent takes the call from different different services but hold music is play astrisk own by default. Is there any way to play different hold music bases on services which run into server A. I have some changes into musiconhold.conf (server B) but problem is no solve. please…

Asterisk Users 3.1 years ago 4 Answers

FastAGI script and DIAL execution

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Hi all, I am trying to control the whole call using a FastAGI script. To that
effect I launch a FastAGI script (written with asterisk-java). Basically, I want to DIAL from within the FastAGI script. When the call
ends I want to control the hangup (if executed at the remote end), and
depending on the cause, dial again, play a message, or hang up. This is a
pretty standard telephony scenario. I did it before by executing the AGI,
setting variables, calling the DIAL command from the dialplan, and then
executing a second AGI…

Asterisk Users 3.1 years ago 1 Answer

IAX Trunk issue.

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I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its extension. It also dials the s extension. I only placed the s extension in the dial plan to verify that the traffic was going across the IAX trunk and hitting the correct context. Any help…

Asterisk Users 3.2 years ago 1 Answer

Tell external number instead of internal number

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Hello, I have an internal extension, e.g. 1005 which is being called from an
external/public number like 123456789. Now when it comes to the spoken
voicemail information it says something like "number 1000 not available",
however it should say "number 123456789 not available". How can I configure
this? I already googled and I guess this is really easy, but I just
couldn't figure out how to do this ): So thanks for any hint :-)

Asterisk Users 3.2 years ago 2 Answers

Digium IP Phones - Teleworker Capability?

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We couldn't see anything about this on the Digium site, but maybe
someone here can comment? Do the new Digium phones provide good "teleworker" functionality? The benchmark we're comparing against is the capabilities of Mitel
3300 IP systems with Mitel 5330 IP phones (running their proprietary
MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select
TW mode and the IP of the gateway server). The phone reboots and it
is ready to be used (once the Mitel border gateway is set to recognize
the unit's ID, based on…

Asterisk Users 3.2 years ago 4 Answers

fax issue

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Hi, I am using asterisk 1.8.9.3 , I am using Spandsp FAX Driver: 20091228
123351, i will clearly explain my scenario. I am sending fax to a anlog fax machine which is connected to mediatrix
analog gateway. say the analog fax extension is 18260.
I have enabled the fax mode , T.38 codec, clear channel codec , cng tone
everything in my analog gateway. while sending a fax from asterisk to the
analog fax extension(18260 this extension is registered in asterisk) , I am
using SendFax dialplan application for sending fax , most of…

Asterisk Users 3.2 years ago 0 Answers

FaX issue

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Hi, I am using* asterisk 1.8.9.3* , *I am using Spandsp FAX Driver: 20091228
123351*, i will clearly explain my scenario. I am sending fax to a anlog fax machine which is connected to mediatrix
analog gateway. say the analog fax extension is 18260.
I have enabled the fax mode , T.38 codec, clear channel codec , cng tone
everything in my analog gateway. while sending a fax from asterisk to the
analog fax extension(18260 this extension is registered in asterisk) , I am
using SendFax dialplan application for sending fax , most of…

Asterisk Users 3.2 years ago 0 Answers

Need queue name in CDR

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Dear All, I am making asterisk report using CDR values given by asterisk. I have queues which consist of multiple members (extension). Also, an
extension may be in multiple queues. So, I want CDR to record the
name/number of queue from which the call was originated. E.g.
*Channel* * DestinationChannel*
* Src* * Destination
*
SIP/KOT-0000000c Local/102@from-queue-6a84;1
0856511524 9999 (first
line in CDR)
Local/102@from-queue-6a84;2 SIP/102-0000000e
0856511524 102 (second
line in CDR)
In above example, 9999 is a queue and 102 is an extension which is member

Asterisk Users 3.2 years ago 4 Answers

Polycom Caller ID

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Hello, I have an issue I remember seeing a while ago and forgot to investigate further. Now it is turning into an issue and will need to be resolved. A customer has Polycom 335 phones (and a couple Soundstation 6000s), and when an extension is calling out, the screen on the 335 shows the company's internal CID number instead of the person they are dialing. This also applies to receiving calls - the internal CID is displayed as opposed to who was calling. I remember seeing something about connectedline issues with Polycom phones, but I can't find the bug I…

Asterisk Users 3.2 years ago 1 Answer