similar to: PJSIP configuration question

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:

Our software utilizes an OS X Server (Snow Leopard 10.6.8) for file read and writes. We gather files out of many directories, process them, and then write files back to the server. We had a long time end user who utilizes our software with an XP box upgrade from XP to Windows 7. Now with Windows 7 the SMBD is crashing routinely under heavy loads. Whatever information it was working

I am not sure what you mean by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP

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

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

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 settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel:

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

<snip> > > I translated those settings to the following for pjsip.conf... > > [transport1] > type = transport > bind = 0.0.0.0 > protocol = udp > > [outbound.vitelity.net] > type = aor > remove_existing = yes > contact = sip:64.2.142.93 at 5060 This is incorrect. The contact should be: contact = sip:64.2.142.93 It will use a default port of 5060. I

This fixed the problem. Developer before me wrote some code to build up the dial string. Always thought that string appeared off, but it worked so I left it alone. Thanks Joshua and George for helping with this. 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 Dan Cropp Sent:

I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? PJSIP is including the Contact for the ACK response to the OK. Contact: <sip:1234 at xxx.xxx.xx.xxx:5060> When using the chan_sip, it does not include that field in the ACK response to the OK. (Been a long couple weeks) Have a great day! Dan

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

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

Hi folks, I'm new to WINE and have actually read the docs, so go easy on me. My machine is setup to dual boot between Red Hat 7.2 and Windows2000. This is what I get when I try to run Eudora 5.1 from my Win2k partition. This is as far it'll go--no window ever displays. I get identical results with either WINE 20020122 or a CVS snapshot from yesterday evening. Let me know if

Hi all, i still have problem.. Some time I am able to install and other not. I tried this on different machine either in a pool and either in a single installation of XCP. These are the steps I use to create a VM: # set disk size xe template-param-set uuid=9d8c3be5-f16b-7446-b978-e23551d8a77e other-config:disks="<provision><disk device=\"0\"

Hi, In continuation with my previous mail, I changed the vif entry in the config file as, vif = ["mac=00:16:3e:43:b7:99,bridge=xenbr0"] and I was able to boot into the guest, but I was not able to ping any system from the guest. I have attached the snapshot of the boot error messages. Thank you Regards Ananth -----Original Message----- From: CB, Anantha Padmanabhan Sent: Wednesday,

Hi, I got this error when starting a domU on Centos 5.1 i386, Device 0 (vif) could not be connected. Hotplug scripts not working. Why is this happening ? I should restart the machine to see if this still happends after a fresh network bridge. But it's somehow a production server, I would like not to restart it, if possible. No domU is currently running. I could start domUs a week ago.