Re: [asterisk-users] CallerID from POTS to SIP

2007-07-29 Thread astuser
Thanks for the reply. Yep, the telco is providing the callerid since it's getting into the asterisk call database just fine, in all cases. It appears that the telco has changed the cadence for the distinctive ring again. I modified the dring contexts to handle a 0,0,0 dring and now callerid

Re: [asterisk-users] CallerID from POTS to SIP

2007-07-24 Thread astuser
Thanks for the reply. Unfortunately that didn't work. What's confusing is that for the line without any distinctive ring that works correctly with callerid, the only thing it does is dial the phones, so here's the entire context: [add-incoming] exten = s,1,Dial(SIP/ht1SIP/ht2SIP/gxp1,20)

[asterisk-users] CallerID from POTS to SIP

2007-07-23 Thread astuser
Hello, I'm having some trouble getting callerid to work through a distinctive ring line. Specifically, I'm having a problem getting the callerid to pass from an ATT POTS line connected to a TDM400P to SIP phones, both hard phones and soft phones. What leads me to believe that it's a problem

[asterisk-users] Voicemail from sip phones

2007-01-28 Thread astuser
Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of

Re: [asterisk-users] Voicemail from sip phones

2007-01-28 Thread astuser
Thanks for the reply. It is there within the sample stuff but it's commented out. ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility

Re: [asterisk-users] Voicemail from sip phones

2007-01-28 Thread astuser
Hmm. Nope. Still same thing. I added pedantic=yes both in the general context in sip.conf and in the user's context in sip.conf with no change. Just for fun, I also changed it to pedantic=no in each place with no luck either. (I stopped and started asterisk between each change). Other

Re: [asterisk-users] Voicemail from sip phones

2007-01-28 Thread astuser
I don't see anything apparent in for the SIP phones that would indicate that *8 is anything but VoiceMailMain(). I looked in extensions.conf for it, as well as sip.conf (the config that I pasted in a previous e-mail. It does indeed appear that a literal *8 is being passed to asterisk.

Re: [asterisk-users] Cordless SIP Phones

2007-01-28 Thread astuser
I share your frustration. Might I suggest a Grandstream HT-386 (or 486, etc) gateway to a regular cordless phone? On Sun, Jan 28, 2007 at 09:52:19PM -0500, Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and

[Asterisk-Users] Asterisk answering machine replacement, WaitForRing(), application return values

2006-04-02 Thread smc+astuser
This is possibly a dumb question, but I've googled around and poked through the documentation and I'm a bit confused. My initial experiment with Asterisk involves setting it up in place of my old dedicated answering machine. That means I've still got a regular old phone on the line which we