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...set, but the display shows 9303551212) It doesn't seem to be digit specific, and can lose one or more digits when the problem happens. Dialing very slow and deliberate seems to help, although I haven't done super serious testing of that yet... Any ideas? Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 303.680.1283 IAXTel: 700.206.7507 FWD: 484162 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040922/eee8eeb5/attachment.htm

Hello everyone.... Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question.... My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\NAT Address I have the public IP of my cable connection. Comcast gives me a

...sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 206.666.1786 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050212/ce088927/attachment.htm

I haven't seen any status on the 7960 backlight project lately...I tried to email the original poster but his mailbox appears to be over quota. Does anyone have an update on this? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 303.680.1283 IAXTel: 700.206.7507 FWD: 484162 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040924/56c7aab7/attachment.htm

...oming and outgoing calls (on one end of the trunk), hopefully resulting in a group count equal to the total of all active incoming and outgoing calls. Does this seem reasonable? If any of you are accomplishing this in other ways, please share your examples... Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 206.666.1786 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050223/cf45f0cd/attachment.htm

...ese phones give me a short status update on any problems or shortcomings that I need to be aware of? I would really like some info on how the IP-10 is working these days with SIP....if it's reasonably solid it's probably at the top of my list... Thanks Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 206.666.1786 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050223/1075c129/attachment.htm

...hort, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation like this (built in switch, display, speaker phone...) Thanks Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 206.666.1786 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050215/830d38d9/attachment.htm

...olved is on the local lan. Lastly, is it possible to use an IAXy? I assume that if you're using ulaw and can minimize packet loss that his could work but I haven't seen much (success or failure) on the list from anyone attempting fax over IAX. Thanks Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 303.680.1283 IAXTel: 700.206.7507 FWD: 484162 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040914/1c18ca2c/attachment.htm

...sages appear in the logs of both servers. It almost appears as though these messages are normal following completion of a call (a hangup), but we are troubleshooting bad audio in both locations and the wording of these messages doesn't appear benign. thanks Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 206.666.1786 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050518/3e1b7d3b/attachment.htm

...ction, I added an entry for the bitmap that I want to display: bitmap.IP_500.66.name ="arf" but from there I'm not sure where to go...what do you change to tell the phone to actually use that bitmap on the main screen during idle conditions? Thanks Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 206.666.1786 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/14b42ec9/attachment.htm

...or if the phone is turned off, the cell will answer the call immediately... If this is not implemented in other channels, can anyone recommend either an alternative solution or a starting point for implementing it in the code of other channels? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 303.680.1283 IAXTel: 700.206.7507 FWD: 484162 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040908/eb339216/attachment.htm

I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: [general] ; format=wav49 maxmessage=180 attach=yes Even if it only gave the caller 30 sec to leave a message it would be nice to tell the caller that they have run out of time before ending the

Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement

...igital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ------------------------------ Message: 7 Date: Tue, 21 Aug 2007 22:20:20 -0600 From: "Marty Mastera" <marty at m3resources.com> Subject: Re: [asterisk-users] Polycom behind NAT won't register to * serverbehind ALG To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <E25AAE8239598347B01E9AB08EEAD0F63B858D at trajan.mastera.us> Co...

> > I've tried setting up my sip.conf in two ways: > > > ------------------------------------------------------ > register => [240xxxxxxx]:[my_password]@sip.broadvoice.com > > > [Broadvoice] > type=peer > username=[240xxxxxxx] > fromuser=[240xxxxxxx] > secret=[my_password] > host=sip.broadvoice.com > context=incoming >

I?ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *,

> I too contacted CDW about the $9.37 Cisco support > contract. But because I did not buy my phone from them I was > not allowed to purchase it. The vendor I bought the phone > from does not provide them. What are the "magic words" to > get CDW to sell it to you? With all of this hassle I highly > doubt that I will buy more Cisco phones anyway. After >

> >> Jason Brown wrote: > >> | Anyone have experiece with polycom phones? > >> | > >> | I am experiencing a really weird problem. In an office > where I have > >> | the following extensions: > >> | On the Polycom phones, when I want to dial from extension > >> 100 to any > >> | extension 120 or above, or dial out, it

We are having an issue with the SPA962/932 combo where the phone and the sidecar will reboot unexpectedly ? could be onhook, could be on a call, doesn?t seem to matter. I read that certain early firmware revisions could cause this so I?m running what was a week ago the newest available (5.1.18). A call to Linksys support suggested that I ensure that the phones are using a recent firmware version

Hello everyone, I playing around with ENUM and have configured * to query a few sources for testing purposes (fierymoon, e164.arpa, e164.org). I'd like to know if there is a way to query these servers manually (ie outside of asterisk via nslookup or equivalent) to find out if particular exchanges are listed with wildcards, so as to terminate calls to those prefixes (I'm not trying to