my thoughts exactly. the issue is, the great keyboard warrior responding to my issue out-rank me by a considerable margin, so i’m having a little bit of a difficulty keeping up with their questions. i have never done a manual config file edit on asterisk, for a reference in where i am. and i appologize for the delayed response, i’ve been having major server issues elsewhere that needed my attention.
you are correct, it’s a NATed PBX behind a router. this can be changed, upon your confirmation that this is causing issues. details: we have never had any ports open for this PBX; our provider has never required it and it’s been running this way for several years. how they are able to call it without open ports is something i don’t fully understand. heartbeats? i can also give this PBX its own public IP address if you recommend it, i would just need to put the appropriate security policies in place for that. including updating our heinous extension credential practices.
this can also be done, if you tell me what you mean by a SIP trace. never done one before. the router is a sonicwall, and like i said, we have not had to implement any SIP related settings on it.
david, to clarify, when using “default” CID setting under FMFM, there is an outright connection refusal. (using the caller’s ID when relaying the call is the desired method) when i’m using “use dialed number” as the CID method, the call connects, but no media is sent. after the configured 30 seconds, the PBX drops the call.
Well, I did ask 9 days ago if you tested with Call Confirmation and got audio when using it. There were a few follow-ups about how that was obviously going to work by others. Yet, you still haven’t followed up and confirmed you tested this and it worked.
my apologies, Tom. that’s what i meant with some of you guys are above my level. after digging around some, i haven’t gotten a perfect grasp on what call confirmation is. but i can say, when using “use dialed number” for CID settings, if the inbound call is to an IVR where you “dial your party’s extension”, the audio works.
when using the “default” CID setting however, the call never connects.
This is classic symptom of a misconfigured router. You need to port forward the entire RTP range (10k to 20k UDP) to the PBX thru the NAT firewall/router. The diversion header is a red herring.
Not for zero-way or one-way audio issues, but you may need to forward it for other reasons such as if your provider sends inbound calls from hosts other than the one to which you’re registering.
ok, with the signaling port 5060 as well as the entire RTP range (10k to 20k as per the settings in the GUI) forwarded i am still at the same place: a directly relayed call, using FMFM has no audio and gets dropped in the 30 second time limit. a “dial your party’s extension” call has audio. this is with “use dialed number” as the CID. I am still trying to get to where i can use the actual CID of the incoming call.
here are the SIP settings to make sure we’re on the same page: (yes i have our correct public IP entered)
as for the router, it’s forwarding 5060 and 10k to 20k as mentioned above. regular calling as well as local inter-network calling has been working fine.
Jared
P.S. as soon as i enable NAT under the Legacy SIP settings, inbound calling no longer works.
OK. There are two issues here. First, your provider is blocking calls coming from you with the wrong callerid. They want a diversion header so they know it is you.
Second, this is an issue with audio flowing and that is related to using non-optimized local channels for Followme. This means SIP (trunk) → local channel (followme) → SIP (trunk) that local channel is staying in the call path. Audio has to flow through it and no ambient noise is flowing from it. The other side might not start streaming audio because it hasnt detected audio coming in.
Why it works with an IVR is because the incoming call is answered which changes how the the call is bridged and audio is handled. There is audio flowing at that point from Asterisk.
Hey Tom, thanks for helping me out with this. I just talked to the provider, and yes, i do need a diversion header so, as you mentioned, the call is processed with a number i’m billed for. he also said they’ve seen this working in the past on their network, so i should be able to get it done.
i have not arrived at that yet: the call is still being rejected due to no diversion header being used. can you help be do that? i assume with the length of this forum exchange it’s not a simple matter?
