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Conference for normal calls
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Help migrating from Italian tiscali provider's router to freepbx and pfsense
sorry in my log there is this response that I didn’t noticed:
<--- Received SIP response (445 bytes) from UDP:213.205.21.8:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 82.84.72.XX:5060;rport=60000;branch=z9hG4bKPj22e297f7-8c97-43ed-99b1-4f1af2082907
To: <sip:0039XXXXXXXXXX@ims.tiscali.net>;tag=ztesiprMnrgvtl*3-7-16648*degh.3
From: <sip:0039XXXXXXXXXX@ims.tiscali.net>;tag=e576f041-1619-43ad-ba3c-dc4a554e0a6e
Call-ID: 90b24124-b24f-4118-83c9-8b5f0f2a7ff8
CSeq: 59478 REGISTER
X-ZTE-Cause: "CSCF-BC005027.BC0056B9.BC0053A1.rmicscf1.ims.tiscali.net"
Content-Length: 0↧
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Help migrating from Italian tiscali provider's router to freepbx and pfsense
theyre responding to this. where’s your initial request that they are rejecting?
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Get new the pack of firmware for ip phone
Hi all.
I’m stuck at how to get newest the pack of firmware that i want to upgrade for ipphone.
I’m using freepbx 13.0.195.1, ipphone: grandstream 1615, my firmware pack: 1.18,
it included firmware version GXP-1610 1.0.4.67 for Grandstream 1615.
I tried with some document but seem like they did not help.
So guys, could you pls help me?
Thank you!
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Help migrating from Italian tiscali provider's router to freepbx and pfsense
I analyzed a new log from the the beginning here is the complete request/response, what I noticed is that I have to place ims.tiscali.net in “from domain” field or I will not receive any kind of response, and I still didn’t understand where to place srvmi.p.ims.tiscali.net
<--- Transmitting SIP request (923 bytes) to UDP:213.205.21.8:5060 --->
REGISTER sip:core1.p.ims.tiscali.net SIP/2.0
Via: SIP/2.0/UDP 82.84.72.XX:5060;rport;branch=z9hG4bKPj7095f76a-c0e7-4596-b530-faf980fa455c
From: <sip:0039XXXXXXXXXX@ims.tiscali.net>;tag=f59c2a58-67d8-4594-8653-bbdcfa0b6e7a
To: <sip:0039XXXXXXXXXX@ims.tiscali.net>
Call-ID: 2635f82e-76da-4ac6-9f53-e3c81a937955
CSeq: 2801 REGISTER
Contact: <sip:s@82.84.72.33:5060;line=vtgejqo>
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Max-Forwards: 70
User-Agent: FPBX-14.0.13.4(13.26.0)
Authorization: Digest username="0039XXXXXXXXXX", realm="ims.tiscali.net", nonce="528d8ab21df7a8e44ac88ec41860c044", uri="sip:core1.p.ims.tiscali.net", response="4437b1537661cc40eb2ecff6f1359059", algorithm=MD5, cnonce="abffcaea-1eff-4e05-be02-2223652e8d49", opaque="aW1zLmNvbS5jbg==", qop=auth, nc=00000001
Route: <sip:ims.tiscali.net:5060>
Content-Length: 0
<--- Received SIP response (447 bytes) from UDP:213.205.21.8:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 82.84.72.XX:5060;rport=60000;branch=z9hG4bKPj7095f76a-c0e7-4596-b530-faf980fa455c
To: <sip:0039XXXXXXXXXX@ims.tiscali.net>;tag=ztesipnIm9aFoj_kP*3-7-16648*eecf.3
From: <sip:0039XXXXXXXXXX@ims.tiscali.net>;tag=f59c2a58-67d8-4594-8653-bbdcfa0b6e7a
Call-ID: 2635f82e-76da-4ac6-9f53-e3c81a937955
CSeq: 2801 REGISTER
X-ZTE-Cause: "CSCF-BC005027.BC0056B9.BC0053A1.rmicscf1.ims.tiscali.net"
Content-Length: 0
[2019-10-04 08:43:03] WARNING[8700]: res_pjsip_outbound_registration.c:993 handle_registration_response: 403 Forbidden fatal response received from 'sip:ims.tiscali.net:5060' on registration attempt to 'sip:0039XXXXXXXXXX@ims.tiscali.net:5060', retrying in '10' seconds↧
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Help migrating from Italian tiscali provider's router to freepbx and pfsense
im guessing this is the second part because you are sending the NONCE in this REGISTER, so hopefully before this you already did this dance
-> REGISTER
<- SIP/2.0 100 Trying
<- SIP/2.0 407 Proxy Authentication Required
-> REGISTER (this is what we are seeing?)
if that’s the case, a few things come to mind, other than incorrect password or username.
the realm is case sensitive and sometimes it has to be something specific and not in your config, like the word “Realm” itself
realm=“Realm”
also you posted username is a full URI not just the phone number portion?
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Extension unreachable / reachable
Hello,
I have one extension on my freepbx who become unreachable and instantly become reachable :
freepbx*CLI>
– Removed contact ‘sip:12@172.16.2.4:5060’ from AOR ‘12’ due to request
== Contact 12/sip:12@172.16.2.4:5060 has been deleted
== Endpoint 12 is now Unreachable
– Added contact ‘sip:12@172.16.2.4:5060’ to AOR ‘12’ with expiration of 3600 seconds
== Endpoint 12 is now Reachable
– Contact 12/sip:12@172.16.2.4:5060 is now Reachable. RTT: 40.226 msec
– Removed contact ‘sip:12@172.16.2.4:5060’ from AOR ‘12’ due to request
== Contact 12/sip:12@172.16.2.4:5060 has been deleted
== Endpoint 12 is now Unreachable
– Added contact ‘sip:12@172.16.2.4:5060’ to AOR ‘12’ with expiration of 3600 seconds
== Endpoint 12 is now Reachable
– Contact 12/sip:12@172.16.2.4:5060 is now Reachable. RTT: 46.113 msec
– Removed contact ‘sip:12@172.16.2.4:5060’ from AOR ‘12’ due to request
== Contact 12/sip:12@172.16.2.4:5060 has been deleted
== Endpoint 12 is now Unreachable
– Added contact ‘sip:12@172.16.2.4:5060’ to AOR ‘12’ with expiration of 3600 seconds
== Endpoint 12 is now Reachable
– Contact 12/sip:12@172.16.2.4:5060 is now Reachable. RTT: 55.394 msec
how can i fix this problem ?
Thanks for your reply 
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Module CustomContext
Nice day ! I’m installed module CustomContext then after i’m create two context mirohost and mirohost1.
How can I include context mirohost1 into the context mirohost tools of the web interface? I can register in the config with my hands
[mirohost]
include = mirohost1
but this is not interesting because there is a web interface. Tell me how please?
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Display CID from inbound calls on extensions not actually ringing
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Outside PBX network calling
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Playing opus records in CDR Reports
how add it?
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One way Audio
Hi,
I ve the problem mentioned in the title.
Here is my setup
eth0---->LAN FreePBX (10.1.1.199)
eth1---->SIP Provider network
Also there are 7 remote sites connected through VPN on the Main site 10.1.1.0/24 where FreePBX is.
I ve a SIP trunk to my provider network which is registered correctly
When i make a call from SIP phone on a remote site to a mobile or another remote phone, in the traces from eth0 in the SDP packets there is an IP that its not on my network (172.16.32.26) and all the RTP packets are going to this IP address.
Any ideas ???
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Addheader to sip call
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Free pbx cti solution
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I finally gave up chan_sip for chan_pjsip and it was fine
For those relying on the Match field for pjsip trunks, know that the Firewall module does not whitelist these IPs automatically as is done for the SIP Server. I have an open feature request for this: https://issues.freepbx.org/browse/FREEPBX-18741
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DISA direct call possible without calling first DISA module?
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I finally gave up chan_sip for chan_pjsip and it was fine
Oh, I’d also check whether you actually do need to specify the transport on the endpoint. Asterisk 16 DNS resolution prefers IPv6, so the resolution should result in AAAA records being used and thus the IPv6 transport chosen. If that’s not happening then I can take a gander and see why.
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Incoming Call and Outgoing both are working but either one works at a time
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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I finally gave up chan_sip for chan_pjsip and it was fine
In FreePBX you have to choose a transport for trunk endpoints. Sounds like another feature request: an “auto” choice that omits the transport= line from the endpoint definition.
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I finally gave up chan_sip for chan_pjsip and it was fine
Ah, yeah, in Asterisk it’s not required unless you have multiple competing transports and want to force it. For most people it’ll choose the right one based on the target of the SIP message.
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