After reading the voipvoip.com site, I'm not sure what format they are after to send calls as relates to pai.
One can assume you are dialing/sending 1+10 digits now. Perhaps asking voipvoip what they "don't like" about the call is in order? ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: Discussion list for users of sipXecs software <[email protected]> Sent: Wed Dec 08 16:10:09 2010 Subject: Re: [sipx-users] Can't place outgoing calls - fast busy Followed this, same result...fast busy. Also get no additional activity in the sipxbridge.log file after your suggested changes. --- On Wed, 12/8/10, Tony Graziano <[email protected]> wrote: From: Tony Graziano <[email protected]> Subject: Re: [sipx-users] Can't place outgoing calls - fast busy To: "Discussion list for users of sipXecs software" <[email protected]> Date: Wednesday, December 8, 2010, 2:40 PM they may be sending the calls in on port 5060, which means the call is not anchored. It also means transfers might not function, etc. No initial "9" is needed. Delete your custom rules. Disable the default local, long distance and toll free rules. Create a custom rule 1+10 digits, send entire number to voipvoipgateway. Place the rule above the AA and voicemail rules, restart services. On Wed, Dec 8, 2010 at 3:36 PM, Victor Williams <[email protected]> wrote: Well, we can receive calls without issue. The only problem is dialing out. Why would the 5080 issue come up if we're receiving calls fine? I've removed all dial plans from the trunk and created another that requires 11 digits be passed after the initial 9. Same thing happens...fast busy. I'm looking at the sipxbridge.log file in real-time at INFO log level and see quite a few messages that look interesting, but don't know how to address them. Anyone want to take a stab at looking at them? --- On Wed, 12/8/10, Tony Graziano <[email protected]> wrote: From: Tony Graziano <[email protected]> Subject: Re: [sipx-users] Can't place outgoing calls - fast busy To: "Discussion list for users of sipXecs software" <[email protected]> Date: Wednesday, December 8, 2010, 2:29 PM I don't see they support registering on a port other than 5060, which is a problem. You should ask if they support you registering on port 5080 and sending inbound calls to you on port 5080. They want "11" digits sent to them, a "9" is not needed. So yes, create a new dial plan accordingly. On Wed, Dec 8, 2010 at 3:10 PM, Victor Williams <[email protected]> wrote: The ITSP is VoipVoip.com. When you state "rule", what does that mean? Dial plan? --- On Wed, 12/8/10, Tony Graziano <[email protected]> wrote: From: Tony Graziano <[email protected]> Subject: Re: [sipx-users] Can't place outgoing calls - fast busy To: "Discussion list for users of sipXecs software" <[email protected]> Date: Wednesday, December 8, 2010, 1:58 PM I would disable that rule and replace it with a custom rule that send "exactly" the number of digits to the itsp that they need. You might use "9" if you are using an analog gateway programmed to accept it. What format does your itsp want you to send? Who is the ITSP? On Wed, Dec 8, 2010 at 2:21 PM, Victor Williams <[email protected]> wrote: Looking for a little direction from some more experienced telephony people...this is my first dealings with setting up any type of phone system in probably 10 years... I've somewhat successfully set up our installation (4.2.1 / latest stable) and can receive incoming calls. We set up with voipvoip.com, and I can call into our local DID and it rings to the extension I set up for all calls to terminate at. However, when I attempt to call out dialing 9 and then the number, I get fast busy. This happens whether I dial 11 digits, 10 digits, or 7. I troubleshot my previous issue with SIP registration by setting the logging level to debug of the "SIP Trunking" service, then tailing the sipxbridge.log file. What would be the best method or best place to start to find out why I cannot dial out? Is there another file that would give some clues by looking at the output? _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 -----Inline Attachment Follows----- _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 -----Inline Attachment Follows----- _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 -----Inline Attachment Follows----- _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
