SIP/VoIP Set up
# support
l
I'm looking to get access to the Vapi SIP trunk for connecting to our current PBX software. I'm just looking to add SIP system and I understand that this is part of the Enterprise plan but I would really like to explore if a one time payment option would be possible as we a really satisfied with vapi. We are open to discuss this matter with the team at a deeper level. I've also already filled out the form and hope to hear back from you soon. Thanks! Best, Aaron
v
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@luisson Hello Aaron, I'm thrilled to hear about your interest in accessing the Vapi SIP trunk for your PBX software. It's great to know that you're considering the Enterprise plan and are looking for flexible payment options. Since you've already filled out the form, our team will be in touch with you to discuss the details of your request and explore the possibilities of a one-time payment option. We're always open to discussing ways to accommodate our clients' needs and find the best solution for you. In the meantime, if there's anything else I can assist you with or if you have any more questions, please feel free to let me know. We're here to help and ensure you have a smooth experience with Vapi. Best regards, Vapi Support Team
You can continue this conversation by mentioning me in the message.
l
Awesome, I've submitted the form a couple weeks ago. just making sure. Thanks!
j
@luisson I already went down this path. They are not open to setting this up as a once off fee. You have to subscribe to all the extra stuff you probably don't need right now to get that one small feature.
s
The good news is that SIP is available to all the pay-as-you-go user as well
so you don't need to be on enterprise plan
just talked about this with my co-founder and they said we can do it for free
Will be uploading this guide to docs
k
This is great news.
What am I missing here. I craeted a sip uri sip:xxxxxxishna@sip.vapi.ai (redacted) using the API as mentioned in the Doc.
when I call it from linphone or zoiper it doesn't work.
@Vapi any help here?
v
@kmskrishna I'm here to help, Aaron! It seems there might be a discrepancy between the documentation and your experience. If you're receiving a 401 Unauthorized error when trying to call the SIP URI, it could be due to a configuration issue or an oversight in the setup process. To assist you further, I'll need to gather some more information: 1. Can you confirm that the SIP URI you created is correctly formatted and matches the pattern provided in the documentation? 2. Have you ensured that there are no typos or errors in the SIP URI you're using to make the call? 3. Is there any additional configuration on your SIP client that might be required, such as specifying a domain or outbound proxy? Once we have this information, we can troubleshoot the issue more effectively. If you've double-checked these details and the problem persists, please ask support for more info on this, as they may need to look into your account specifics or the API's behavior to resolve the issue.
k
Did everything you asked still the issue is there
@Sahil for your reference.
s
Temporarily we have disabled the auth so you can use any username and pass
it will work.
k
@User Got it. Did you do it just now? Also how stable is the SIP server? Can we run production systems on that?
s
Yes, you can run on the production system
yeah we did like on sunday
we will expose a new way for authentication part
k
because I noticed that sometimes I am getting a 503.
like randomly. From last 2 days it didn't work and today it started working. with no changes from our side
and also just few min back it didn't work. now it is working.
@Sahil In all these cases I didn't do any configuration change. Where as calls to thetestcall@opensips.org is working fine.
What might be the issue?
s
not sure man
do you have pcap?
i can look into that
m
Unfortunately, the problem with the missing authorization still exists. However, if you add a set Authorization header to the first SIP INVITE, it works. It seems that your SBC, proxy or so requires an Authorization Header but does not check further. During operation, however, I noticed that calls are always terminated after 30 seconds due to silent timeout. I wonder if people are talking and answering all the time.
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