Detecting Fax Tones over IAX2

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Hello All,
I use IAX2 as the incoming connection from my DID provider. For whatever
reason, this works best for me, SIP connections lag very frequently and
only have about a 50% success rate for incoming calls (they get dropped
mysteriously). I'm trying to implement a fax/voice switch. I have faxdetect=both in my
sip.conf, and when I use sip, it works well. However, from what I can
tell, there's no such option for IAX2 connections. Any ideas on what I can do here, or am I out of luck? Thanks,
Cody

Asterisk Users 3.3 years ago 3 Answers

Unable to execute 'dahdi_scan > /etc/asterisk/dahdi_scan.conf'

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On Wed, May 23, 2012 at 02:02:51PM +0500, p070075 Muhammad Atif Ramzan wrote:
> Hi
>
> Can anyone help me with this error
> Unable to execute 'dahdi_scan > /etc/asterisk/dahdi_scan.conf'
>
> i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call
> reached the destination but no voice is coming from destination my voice
> reflects back Have you verified the user asterisk is running as can execute
dahdi_scan? This was asked not too long ago on the forums as well: http://forums.asterisk.org/viewtopic.php?f=1&t=82659

Asterisk Users 3.3 years ago 0 Answers

twenty thousands (20, 000) users, which asterisk and how many servers

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the solution lies in kamailio/opensips's despatcher module. Sent from my iPhone On 23 maj 2012, at 20:46, bilal ghayyad wrote: > Dear;
>
> So it is a hardware issue and not software?
> I am afraid that asterisk software it self is not able to support 20 000 users and 2000 concurrent calls.
>
> About the high availability: is there a method that if the first asterisk server down, then the call will stay connected and failover to second asterisk server?
>
> Regards
> Bilal
>

Asterisk Users 3.3 years ago 5 Answers

SIP endpoints CANCEL when PRI receives Cause Code 31

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We have an Asterisk server which connects to another Asterisk server
acting as a PSTN gateway. This gateway machine has Digium TE210P card
connected to a pair of PRIs. For the most part, all is working well, however there are some specific
telephone numbers that my users have attempted to call, but we unable to. I set debugging on and determined that when the the gateway machine
dials one of the numbers in question, we receive from the PSTN an ISDN
cause code 31, which in my understanding is not an error. This is then

Asterisk Users 3.3 years ago 2 Answers

No caller id when using cadence with DAHDI

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Hello everyone, Just thought to let you know of a weird issue in Asterisk 1.8.? + Dahdi
2.6.? (and 2.5.?). When you specify any cadence in an app (Dial, Queue) then caller id does
not work. For instance with the default cadences (everything commented out in
chan_dahdi.conf) : Dial(DAHDI/54) caller id works Dial(DAHDI/54r1) caller id does not work (even for r1) I just found this issue did not have time to investigate further. Can
anyone else verify that this is true for tonezones other than 13 (gr) which
I am using? Cheers, Panos

Asterisk Users 3.3 years ago 1 Answer

Vitelity Setup

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On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N
wrote:
> Hi,
>
> I am unable to register vitelity SIP trunk, where its keep on sending
> registration request, and I am using Asterisk 1.4.39.2, my registration
> procedure as follows,
>
> sip.conf
>
> register => username:secret@sip41.vitelity.net:5060
> We use viteity w/o registration like so: [vitel-inbound]
type=friend
dtmfmode=auto
host=inbound24.vitelity.net
context=vitelity-inbound
allow=all
insecure=very [vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=vitelity-outbound
allow=all
insecure=very

Asterisk Users 3.3 years ago 12 Answers

twenty thousands (20, 000) users, which asterisk and how many servers?

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Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use…

Asterisk Users 3.3 years ago 2 Answers

Transfer call issue

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Hello, a client attempted to transfer a call today which failed and returned the channel back to her. When this happened on the console we saw: Got OK on REFER Notify message the version that we are running is 1.8.9.2. Are you aware of any none issues please with this version as I could not find anything in Jira ?

Asterisk Users 3.3 years ago 1 Answer