Archives : March-2019
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I have 2 PBXs, one in each office (say one in New York, one in Boston).Ihave mobile users that can show up at either office and connect their soft phones. Is there a very simple DUNDI config available which describes how to set this up?Also, can I h..
Does anyone know if there is a way to disable the norefersub for PJSIP?It appears this is causing problems with a test were running with Cisco.A wireshark trace from a system where the transfer with Cisco works versus a trace with Asterisk/Cisco sh..
I am currently not using qualify, but it seems like a nice way to know if the phones are online.I attempted to set it up, but am running into a 404on the subscription.1.From the manager, Action: PJSIPNotify (with an endpoint).This caused the follow..
Mike In rtp.conf, what are the port ranges you specify? I had almost exactly the same problem not too long ago. People will phone, and sometimes it will work, sometimes not – one way audio would happen, then start working, then stop working. The prob..
–_000_46FCA36BC7E1D546AC28870158F9C25B03D4024CE3ABRAHAMhighpo_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableDoes anyone have an (overhead) paging system that they like that works with SIP?Weve got a client w..
Hi. Im reading https://wiki.asterisk.org/wiki/display/AST/Function_CDR and wondering what r – Searches the entire stack of CDRs on the channel. means. Can anyone help with understanding this? Thanks, Antony. — Normal people think If it aint broke, d..
I know this was discussed years ago – but Im looking into whether things have changed.Imagine this scenario: 1. Phone A call Phone B through Asterisk.(A — > Asterisk — > B)2. All 3 devices have public IP addresses, and Asterisk is configured for directme..
We have a couple asterisk11 servers behind a Kamailio4 proxy. We are in the process of upgrading to asterisk16 and Kamailio5 and Im testing out Path:support with chan_sip (migration to PJSIP is not possible right now due to integrations with other systems).Functionality-w..
all,I have a user who is reporting one-way audio, but only when a call is made to or from particular PSTN (cell) numbers.Their phones are behind a NAT router and my server is on the open Internet.Calls within their office sound fine.Calls to/from m..
I was trying to find if Asterisk supports Dante ?Dante — https://www.audinate.com/AES67 — http://www.aes.org/publications/standards/search..