similar to: PJSIP configuration question

Thanks George. That was the ip address I was given. Unfortunately, my contact at Vitelity is gone for the day so I can?t verify it with him. I added the qualify_frequency as you suggested and it does appear that I have something configured incorrectly?. <--- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 ---> OPTIONS sip:64.2.142.93 at 5060 SIP/2.0 Via: SIP/2.0/UDP

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*

On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp <dan at amtelco.com> wrote: > Not sure why, but Vitelity changed the settings to IP based authentication > on me. Here's the new sip.conf settings they sent me. > > type=friend > dtmfmode=auto > host=64.2.142.93 > allow=all > nat=yes > canreinvite=no > trustrpid=yes > sendrpid=yes > > When I use these

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > You need the network and mask. For example if the ip

On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at amtelco.com> wrote: > > Yes, everything is behind the same NAT. > > > > For the application I?m working on, the only endpoint is the endpoint to > Vitelity. > > We use AMI to Originate calls from Asterisk endpoint through Vitelity to > phones. > > After that, we control the call through AMI to perform the

On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan at amtelco.com> wrote: > > Hi George, > > > > Thank you for looking into this. > > This is behind a nat? > > > Just to be clear...both the pbx and local endpoints are behind the same NAT? > [global] > > type = global > > debug = yes > > > > [transport1] > > type = transport

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at amtelco.com> wrote: > > Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0. > > > > Same problem is happening with both of them. > > > > Could this be caused by PJPROJECT 2.3? > > > > Anyone have any suggestions for what I can try? > > > > My boss is giving me until

Here's an update... My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have. He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net At this point, it seems to be working (and this is going through a Cisco

Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From:

On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com> wrote: > > Thanks George. > > I will correct my local_net in the morning. > > Vitelity chan_sip settings I have working, do not have a fromuser. > sip.conf settings... > > I think you can actually specify anything, it just has to be populated with something other than a sub-account username. >

Dan Cropp wrote: > I corrected my local_net setting (based on advice from network admin). > > I have tried several different values for the from_user and still have > the same problem. > > Asterisk receives the OK from Vitelity. > > Asterisk sends the ACK (without a Contact header). A Contact header is not required to be in the ACK. > > Vitelity doesn?t seem to

Thank you Joshua. I will make the modifications this morning and give it a try. Have a great day! Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, December 10, 2014 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP

I am running PJPROJECT 2.3 and Asterisk 13.0.0. I answer the call, about 15 seconds later, vitality hangs up on my cell phone. However, Asterisk is never notified When the OK (for the answer) occurs, the ACK seems to never be accepted. The OK recvd with ACK sent occurs several times. Here are the pjsip.conf settings... [global] type = global debug = yes [transport1] type = transport bind =

Ok, it didn't quite solve everything. There is one slight issue. When I answer the call on my cell phone, Asterisk sees it as answered. I can play audio, send dtmfs, etc and hear it on my phone. However, a short while later, Vitelity tears down that call and Asterisk is never notified about it. I tell Asterisk to hang up the call (via AMI) and it is removed from Asterisk. I gather the

I am trying to configure a connection to BluIP. I am able to make incoming calls work. However outgoing calls are not working. For the Outbound Registration, I noticed the contact field is always the internal IP address of my pc instead of mycompany dot com I can Originate (using AMI) to my Vitelity trunk (IP based authentication). However, when I Originate to my BluIP, it is being rejected.

Trying this again after my first away from work in a couple weeks. Running Asterisk 13.0.0 IP authentication with Vitelity I can Originate with sip, but not pjsip. Here is the sip settings and trace. Action: Originate ActionID: S8 Channel: SIP/8005555555 at outbound.vitelity.net Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable:

Thanks George. I will correct my local_net in the morning. Vitelity chan_sip settings I have working, do not have a fromuser. sip.conf settings... [HVout] type=friend dtmfmode=auto host=64.2.142.93 disallow=all allow=ulaw canreinvite=no trustrpid=yes sendrpid=yes nat=yes context=TestApp On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph at fairview5.com<mailto:george.joseph at

I am not sure if I entered the correct settings for the transport information. For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered. One minor detail, we are using ip authentication. When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for

I corrected my local_net setting (based on advice from network admin). I have tried several different values for the from_user and still have the same problem. Asterisk receives the OK from Vitelity. Asterisk sends the ACK (without a Contact header). Vitelity doesn?t seem to process it, so they send an OK again. The OK receive, Transmit ACK occurs 4 times. A short while later, Vitelity hangs up

Yes, everything is behind the same NAT. For the application I?m working on, the only endpoint is the endpoint to Vitelity. We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones. After that, we control the call through AMI to perform the work we need. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of