Thread: RE: Google Voice Peering |
dji
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As on normal sip trunks, one can specify the outbound caller ID. This does not appear to be an option with the Google Voice add-on.
Is this function that can be added?
djino
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Thread: RE: How To Change a number from what is dialed to what is sent to trunk |
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So far there are only 3 310 numbers that I want to change. But will add more when I see the need for it.
1) Someone dials 310-7873 should be changed to 1-800-773-2121
2) Someone dials 310-2355 should be changed to 1-888-333-2811
3) Someone dials 310-7777 should be changed to 819-682-1515
Thats what I'd like to setup so far and add more when I see other common 310 numbers being dialed.
And yeah, its a complete digit replacement, which I'm not sure how to implement. Any idea?
djino
"Thanks for your help"
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Thread: RE: How To Change a number from what is dialed to what is sent to trunk |
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Thats what I thought I'd have to do, but the thing is, the 310-xxxx number won't be the same as 1-800-310-xxxx. The 1-800 number will not have any string of digits in common with the 310 number.
I assume when you put 1800+310xxxx, that means if someone say dials 310-1234 that it will pass to the trunk as 1-800-310-1234.
But what I want is completely different number to be dialed. Example: someone who dials 310-1234 would pass to the trunk as 1-800-987-6543.
Is this possible?
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Thread: RE: How To Change a number from what is dialed to what is sent to trunk |
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No No no, I'm not speaking about Caller ID at all.
My ITSPs are not able to dial 310-xxxx numbers. I would like for whenever someone makes an outbound call to 310-7873, that pbxes changes this to it dialing 1-800-123-4567 for example on the trunk I specify.
Is this possible?
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Thread: RE: Unable to receive inbound calls on Callcentric iNum route |
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I've recently requested and received a iNum through Callcentric.com.
I then added a callcentric trunk to my pbxes setup along with creating an inbound route.
I'm not sure what to call the inbound route name. I've tried giving it the full 0118835100xxxxxxxx iNum route name, I've also tried giving it the 8335100xxxxxxx iNum route name and even starting with 5100xxxxxxx iNum route name. Last thing I even tried is using my cellcentric username as the inbound route name.
In all cases above, whenever someone calls the iNum above, my extensions don't ring.
I check the Monitor log, and it shows the log of someone attempting to call, but pbxes doesn't seem to find the inbound route as it shows the letter "s" under destination (when I've told it to route the inbound request to an extension on the pbxes system).
Please help. Thanks.
djino
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Thread: RE: Caller ID does not pass |
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Zitat: |
Originally posted by Diafora
There is a way, but if and only if the ITSP you are using for the outgoing call to your cell phone, allows you to set the CallerID.
There are several threads in this forum discussing this issue. |
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Hmmm... Ok, and if the ITSP allows me to set the Caller ID, how would I go about fixing that?
I've been searching the forum, can you point me in the direction on where this has been discussed?
djino
"Thanks!"
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Thread: RE: Cannot register on any of my SIP Extentions |
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Zitat: |
Originally posted by Diafora
On which server is your account hosted? (i.e. what is the number X in the wwwX.pbxes,com URL?) |
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4
Zitat: |
Have you tried a Submit & Start from the Personal Data section |
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Yes, still doesn't work
Zitat: |
if you have access to the web interface of your account? |
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Yes. I can make changes in the web interface, but none of my extensions are able to register. I can't register via X-Lite on my system, nor with Fring app on my cell phone, nor with my home voip terminal SPA2102. All fail to register when using pbxes sip creds.
djino
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Thread: RE: Cannot register on any of my SIP Extentions |
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Is anyone else having the same problems today?
I have about 5 SIP extensions with pbxes and neither of them are able to register to pbxes.
I've also tried to call the 2 DIDs associated with 2 trunks via my cell phone, but it just keeps ringing.
djino
"I really hope this can get resolved ASAP!"
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Thread: RE: Caller ID does not pass |
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I have a trunk setup with an inbound route. Someone calls my Home DID, the inbound route for this then sends the call to my cell phone PSTN extension.
When the event occurs that someone calls my Home DID, my cell phone rings and shows my Home DID as the caller ID.
Is there any way that pbxes can pass the Caller's Caller ID, instead of my Home DID Caller ID?
djino
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Thread: RE: SIP Inbound calls & SIP URI destination issues |
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Zitat: |
Originally posted by Diafora
After a few iterations the call to the Gatineau number ends up on a Voice Mailbox. I am not sure where it resides though. |
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Yes, I saw when you made that call. That is the correct outcome, as you heard it ring several times then a voicemail picked up.
The Gatineau DID is fine, I never had issue here. The only time this was an issue is when 1) I edit this Gatineau Inbound route for it and tell it to go to a SIP URI.. and tell it to send the call to an outside SIP Address. I get a busy signal here. Its as if the SIP URI destination option alone for pbxes does not function as it does not dial-out to another SIP Address that is placed there.
2) Besides that issue, the other main issue as described in the original post is when using callcentric.com SIP (with X-Lite application), I will dial 11111727111@toronto.voip.ms , this is a Virtual SIP that gets forwarded to my Voip.ms account (the 111727 account @ toronto.voip.ms), but pbxes does not answer the call when there is more than 1 inbound route.
If I have just 1 inbound route, the SIP inbound call rings the extension I have set. But if we create 2 inbound routes (one for the Gatineau DID, and another for the SIP DID, then pbxes can't seem to find the VoipMS inbound route, so it just returns a busy signal.
djino
EDIT: Actually 1) works now. I had to change the inbound route name from Voip.ms to 11111727111
Now when someone from outside calls 11111727111@toronto.voip.ms , it will ring the extensions listed for that inbound route.
2) This issue is still pending (not working). ie. If I were to tell it to forward an incoming call to a callcentric SIP, the SIP URI destination function doesn't appear to be working.
EDIT: Found a work around for #2.
djino
"I'm good to go now, thanks for your help!"
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Thread: RE: SIP Inbound calls & SIP URI destination issues |
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Diafora,
The CyInAc one is a real DID that can be accessed by PSTN. The Voip.ms DID, is only SIP.
They are both real DID. The Voip.ms is only Virtual in the sense that it can only be accessed via SIP. The CyInAc (cybersurf Internet Access) is a PSTN DID with my Local Gatineau, Quebec number (as you can see when you look at the username).
Yeah, you are right about the dial-pattern. Just not sure how I can configure a dial-plan that only includes US/Canada calling.
djino
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Thread: RE: SIP Inbound calls & SIP URI destination issues |
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Hi Diafora,
Even after you've made the changes to my account (changing the trunk names/etc), I still get a busy signal when I call 111111727111@toronto.voip.ms from another sip account (i.e. callcentric.com)
Even calling djino@pbxes.org gets a busy signal.
FYI: the above voip.ms account is the Voip.ms virtual DID, that gets forwarded to the Voip.ms account I have a trunk for.
Again note, this all works perfectly when I reduce it all to 1 inbound trunk that is left blank (for the inbound trunk name).
djino
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Thread: RE: SIP Inbound calls & SIP URI destination issues |
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I have been struggling with my setup.
I currently have 2 Trunks that I use both for inbound and outbound routing.
1) Cybersurf Internet Access - Local DID number - Trunk name = CyInAc
2) Virtual SIP DID - Trunk name = VoipMS
If I setup the following only using 1 inbound route for all calls, I can receive calls successfully on both CyInAc DID and VoipMS Virtual SIP DID.
But as soon as I make any of the following changes, I don't get the result required:
1) Keeping 1 inbound route, but choosing the SIP URI destination, where I input a SIP URI of a place that is known to receive inbound sip calls, the result is getting a busy signal when an inbound call is received on either CyInAc or VoipMS DIDs.
2) From above, I change SIP URI back to ringing an extension when an inbound call is received, but this time instead of keeping the inbound trunk name blank, I change it to Voip.ms . The result is also getting a busy signal when calls are received on the VoipMS Virtual SIP.
What I would like to accomplish in the end is having 2 Inbound routes. If someone calls the CyInAc DID, then calls will ring an extension, if someone calls the VoipMS Virtual SIP during regular hours, then the call goes to a Digital Receptionist, after hours then the call should be sent out to a SIP URI. But due to the above 2 issues, I cannot get this to work and cannot figure out why. Any help/responses would be appreciated.
djino
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Thread: RE: 3 Different Digital Receptionist at 3 Different Times |
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What I would like to know is how I would setup the following:
Each day I want:
1) 12midnight to 8am - Digital Receptionist #1 is used.
2) 8am - 4pm - Digital Receptionist #2 is used.
3) 4pm to 12midnight - Digital Receptionist #3 is used.
All of the above have to be set coming off of the same inbound trunk.
Is this possible? How?
If I only needed two Digital Receptionist, I could easily set Regular Hours and After Hours options to satisfy this. But with 3 DRs that need to play at 3 different times on the same inbound trunk, I'm not sure how to implement if its possible.
Thanks.
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