Check for any kind of SIP interference from the end user's router. Thanks, --Warren Selby, dCAP
On Oct 14, 2011, at 2:38 PM, Adam Robins <arob...@pharmacentra.com> wrote: > Thanks I will do that. The user is remote, so I must first RDP into her home > network and do it from her PC. > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas > Sent: Friday, October 14, 2011 3:35 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone > > I use 501’s here and I can pull up the settings by typing > http://1.2.3.4/index.htm - where 1.2.3.4 is the IP address of the phone. If > you can do that, perhaps something there will be of use to you. > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins > Sent: Friday, October 14, 2011 2:26 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone > > Turned on “sip set debug peer 1234”. I see the qualify messages. I see when > she calls me on my internal extension. I see no SIP messages at all when she > calls my cell phone. > > I understand what Doug and Eric are saying. I need to get into the phone’s > web interface to see how it is programmed just to validate that the phone is > still as I programmed it. What is strange is: > > a. Phone “A” can dial local extensions but not external, so I send her > Phone “B”. > b. Phone “B” cant dial outbound at all > c. Both phones were successfully tested for both call types prior to > shipping and were not in any way reconfigured subsequent to testing. > d. I have not modified the digitmap is sip.cfg in years, and even so, > entering the number and then pressing ‘Dial’ doesn’t work either. > > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind > Sent: Friday, October 14, 2011 2:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone > > Hey, > Can you enable sip trace for that particular sip extension. This sounds weird > that while other INVITES from the phone are reaching but the external > extensions are filtered. If there are no invites for external calls only then > more chances are that the phone is using some dial pattern(phonebook help) > etc like Doug and Eric said. Sometimes in asterisk console I don't see > anything in logs if the Sip extensions' context don't contain the number that > is being dialled > > Do you've access to any phone debugging console? > Sounds like problem is somewhere around "She" :p j/k . > > -- > Regards, > Sammy. > > On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins <arob...@pharmacentra.com> > wrote: > The phone was originally provisioned from an FTP server when it was inside > our network. Once in the field, the phone no longer has access to that > server (it could if I wanted it to). It boots using the last known config, > which worked before shipping. I've been doing it this way for 5+ years. > This is the first problem of its kind. I can get into the phone by RDPing > to the users laptop over VPN and then accessing the phone web interface. I > will try that. > > Please remember, I've already tried two phones, both of which worked fine at > another remote location prior to shipping, having been programmed from good > config files. The first one actually worked fine at this remote location for > a period of time and then suddenly "went bad". > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling > Sent: Friday, October 14, 2011 1:16 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone > > I am assuming you are using a provisioning server. > > If the phone is running firmware 3.2 or earlier you can access the phone web > interface and confirm the dialplan active on the phone is the same as what > you set in the config file on the server. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins > Sent: Friday, October 14, 2011 12:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone > > I've already done that. Both phones worked fine in a different remote > location just prior to shipping. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle > Sent: Friday, October 14, 2011 12:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone > > > Adam Robins wrote: > > No change, thanks > > Well, > > In the long run, it may just be easier to send her out a replacement phone > and ask for that one back, so you can test in house. > > Doug > > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? 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If you are not the intended recipient, > please contact the sender by reply email and destroy all copies of the > original message. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > The information contained in this transmission may contain privileged and > confidential information. It is intended only for the use of the person(s) > named above. If you are not the intended recipient, you are hereby notified > that any review, dissemination, distribution or duplication of this > communication is strictly prohibited. If you are not the intended recipient, > please contact the sender by reply email and destroy all copies of the > original message. > > The information contained in this transmission may contain privileged and > confidential information. It is intended only for the use of the person(s) > named above. If you are not the intended recipient, you are hereby notified > that any review, dissemination, distribution or duplication of this > communication is strictly prohibited. If you are not the intended recipient, > please contact the sender by reply email and destroy all copies of the > original message. > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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