![]() ![]() Telecomunicaciones Abiertas de México S.A. The pjsip configuration is a little more complex as the channel driver's architecture allows for more flexibility in configuration, so things tend be more modular and broken out. Is there a way to identify the endpoint that is causing these messages to appear so I can configure it properly? ![]() WARNING: chan_pjsip.c:712 chan_pjsip_write: Can't send 10 type frames with PJSIP I am getting flooded with these messages: In general chan_sip seems a very robust and reliable technology that can recover easily from any network disturbance, pjsip quite the opposite, I really don’t look forward to the day when there is only one choice, and that is pjsip.įor now I have been forced to stick with chan_sip.On Tue, at 11:29 AM, Carlos Chavez wrote: With chan_sip this setup works fine, although a little inefficiently. A user at location one has a iPhone app that logs into a pjsip extension to check voice mail, this works fine from any location except location two when the registration is matched with the trunk “match” IP field, sent to the trunk not the extension to register and fails. Configure FreePBX Sip Trunking with PJSIP IP based. One other strange thing I have noticed is I have two pbx’s (one and two) with a pjsip trunk working fine between them. PJSIP Softphone Extensions Won't Hangup (Zoiper) General Help pjsip stevensedory (Steven Sedory) February 16, 2018, 3:59pm 1 We’re on: FreePBX 13.0.192.19 Asterisk 13.17.2 We’re changing over from chansip because we continue to have Asterisk crashes when in our preferred configuration: TCP and non standard port for endpoint connectivity. Configure FreePBX Sip Trunking with PJSIP IP based Configuration. The match field only seems to update on an asterisk service restart. This softphone has been tested and shown to be stable in Windows, Linux and OSX. Interestingly I tried to set the match field to allow all IP’s and this caused the whole pjsip channel, all trunks and extensions to stop working. ZoIPer Communicator is a multi platform softphone offering a free version with many useful features. Remote pbx’s only make a partial recovery, they update the dns name fine and can call into the pbx again on the new IP but fail to update the match field to allow calls from the new IP to come in. Firstly the on pbx behind the IP change the pjsip channel collapses completely, no trunks are available and interestingly even pjsip extensions become unavailable and only recover after restarting the asterisk service. However on pjsip this has proved a nightmare. With chan_sip a change of IP at any location can quickly be recovered from with all trunks, service providers and internal links back up and working in less than five minutes without any need for user intervention. My biggest problem has been dealing with dynamic IP addresses. My problems with these trunks remain as I haven’t had time to investigate further.Ī trunk between two Freepbx 13 systems is working fine. I have another trunk with orbtalk (UK Supplier) that requires an out bound proxy, that also registers fine with pjsip, calls can be made in and out but I only get audio in, no outbound audio on external calls. I have a sipgate (UK Supplier) account setup with pjsip, it registers fine, will receive calls fine but refuses to let me dial out. ![]() I have been experimenting with pjsip on both freepbx 12 and 13 with various success and failure.įrom the point of extensions there seems to be no difference, chan_sip and pjsip have worked well for me, the benefit of multiple end points on pjsip is useful.Īs regards trunks I have had a lot more problems. ![]()
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