Sip Can Not Transmit Fax Receive From Chan Dahdi


hello every body

i have problem in receiving fax from e1 lines. this is my scenario:

faxphone----ericson pbx ---e1----asterisk----sip-----zoiper-softphone

when i send fax from zoiper, i can receive it successfully on the faxphone but when i send fax from faxphone, i can not receive it on zoiper. i want do this just by using faxdetect option. when fax comes from faxphone, chan dahdi on asterisk detect fax and redirect to fax extension. in fax extension, i just dial sip peer which is connected to zoiper like this:


peer-1 is a sip peer which is defined in sip.conf like this:

[peer-1] host2.168.0.XX type=peer context=from-trunk insecure=port,invite

i set…

Asterisk Users 11 days ago 0 Answers

DTMF Issue


Hello folks,

We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance.

I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets.

I have enabled DTMF logging and spoken to the SIP provider, but they…

Asterisk Users 26 days ago 9 Answers

Can Dial Plan Handle New Proprietary SIP HEADER Fields? How?


Dear asterisk-users,

I have listened that a diaplan on Asterisk can extract information from proprietary SIP messages header fields. That is, if Asterisk receives a SIP message with a modified HEADER (containing proprietary fields) , is it possible to program the dial plan to make Asterisk extract the values of such fields, being possible to handle such values in diaplan, isn't it?

If it is true, is it also possible to use dial plan to make Asterisk include proprietary SIP HEADER fields in a specific SIP message?

Any hint will be very helpful!

Best regards.

RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55…

Asterisk Users 2 months ago 0 Answers

Am I Cracked?


Hi list!

Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message:

== Using SIP RTP CoS mark 5 -- Executing [000972592603325@default:1] Verbose("SIP/", "2,PROXY Call from 0123456 to 000972592603325") in new stack == PROXY Call from 0123456 to 000972592603325 -- Executing [000972592603325@default:2] Set("SIP/", "CHANNEL(musicclass)

Asterisk Users 2 months ago 19 Answers

Chan_ooh323 To Sip , No Connected Line Info



We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323.

Connection can be shown as


When I do call as avaya user I see name of remote end avay user, i.e. connected line info.

As I see in debug remote side send is as

14:07:29:758 Received H.2250 Message = { 14:07:29:758 protocolDiscriminator = 8 14:07:29:758 callReference = 47 14:07:29:758 from = destination 14:07:29:758 messageType = 7 14:07:29:758 Display IE = { 14:07:29:758 Disa 14:07:29:758 }

over h323.

But now we need to connect to another asterisk over SIP.

in this case we have…

Asterisk Users 3 months ago 1 Answer

What's The Best Average Duration For A SIP Test Call?



What is your experience: if you plan to make a test SIP call to check voice quality overa connection what would be the best call duration? The point is that we should have a call long enough to be able to catch/hear impairments that the connection may have.

This is partly a matter of curiosity, but I believe the roots of this question may be quite important.

Thanks! Valeri on behalf of Sevana

Asterisk Users 4 months ago 0 Answers

Anonymous SIP Calls


We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by netblock filters. We have the usual firewall and fail2ban intrusion prevention and detection set-ups in place.

Our connection to the rest of the world is via PSTN.

We do our own DNS, both forward and reverse. We have NAPTR and…

Asterisk Users 4 months ago 7 Answers

Asterisk On OpenWrt (first Time User)


Hello list,

I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use.

I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP provider for a PSTN connect.

My Ekiga test calls are successful. So it does seem to work :)

The router is configured like this:

- has a user 'asteriskpbx' so Asterisk doesn't run as root - has a USB…

Asterisk Users 4 months ago 0 Answers

Asterisk Only Registering At One Provider



I am running default Asterisk 11.16.0 on a FreeBSD-Machine. I need to register to several other SIP-Services (actually 3):

short sip.conf

register => XX@a register => XX@b register => XX@c

If I remember correctly this worked quite well, but I now checked the system again and it is only obeying the first register statement. "sip show registry" only reports the first entry and if I reorder them, this effect stays the same.

Did something changed recently in the parsing code for sip.conf or so?


Asterisk Users 5 months ago 2 Answers

Switching From SIP To Skype..or Not


I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok.

Is swapping out SIP for Skype a big deal?

Heh, well, I guess it's dead:

If I have a really bad connection, can I "downgrade" SIP somehow? I don't really need to use to make voice calls. Or, more specifically, quality, echo, distortion aren't relevant. Just SIP to SIP "hello".

When I connect to any SIP provider, ekiga, etc, without using Asterisk, I get "too many hops" errors. While I have another computer on…

Asterisk Users 5 months ago 10 Answers