Call Forwarding In Asterisk


Hello Group,

I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue.

Please help me.

I have tried calling with two SIP end point forwarding , even that is not working,

My dial plan line is , Dial(SIP/19201/19202,300)

Asterisk Users 1 days ago 4 Answers



I am trying to set add a SIP Header to a call before adding it to the Queue.

The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs (verbose 999). In the SIP case, I see it.

Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the call when the Queue…

Asterisk Users 11 days ago 4 Answers

Incoming Calls Get 488 Error


I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an "488 not acceptable here". From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as first choice.

I have a SIP trace below. Can someone suggest why the 488 is being generated?


Asterisk Users 14 days ago 2 Answers

Asterisk RealTime Sippeers, Rtcachefriends=yes, Phones Lose Registration


Hello, we have an issue where after a couple of days, a few random phones will lose registration. I don't notice any particular pattern. Out of 200, only about 5-10 will be suffering at any given time and we won't know until the user complains they are not receiving calls. "sip show peers" does not show the phone in the list. I see packets coming into the server, but asterisk is not responding. Rebooting the phone to signal another REGISTER, yields no results. It seems Asterisk simply ignores the packets. I don't see anything in the logs that are relevant…

Asterisk Users 25 days ago 0 Answers

SIP Phones Over VPN Drop Audio One-Way


I am running Asterisk 11 on CentOS 6.x using the DAHDI module with 8x PSTN analog phone lines for outside connectivity. Internally, I am using several models of Yealink SIP phones (e.g SIP-T32G) on a dedicated VoIP network, I have a few of these Yealink SIP phones configured with an OpenVPN certificate so that users working remotely can directly access the phone system (VPN subnet is Note that this is not a NAT; VPN clients are able to directly address the Asterisk server and other SIP phones. Last week the phones connecting over the VPN started…

Asterisk Users 1 months ago 0 Answers

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 2 months 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 2 months 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 3 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 3 months ago 19 Answers