Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Eric "ManxPower" Wieling
Answer() is seldom the solution. Rob Hillis wrote: > Steve Totaro wrote: >> If you ever have problems with a call dropping after 30 seconds, >> Answer() is usually the cause. >> >> > > Answer is the /cause/? Or do you mean it's the solution? -- Consulting for Asterisk, Polycom, Sangoma, Di

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Rob Hillis
Steve Totaro wrote: > If you ever have problems with a call dropping after 30 seconds, > Answer() is usually the cause. > > Answer is the /cause/? Or do you mean it's the solution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.c

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Brent Davidson
Steve Totaro wrote: > On Wed, Jun 11, 2008 at 2:44 PM, Brian J. Murrell <[EMAIL PROTECTED]> wrote: > > If you ever have problems with a call dropping after 30 seconds, > Answer() is usually the cause. > > Thanks, > Steve T This is the first I've heard of this. I've never actually had the drop

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Jared Smith
On Wed, 2008-06-11 at 14:51 -0400, Steve Totaro wrote: > If you ever have problems with a call dropping after 30 seconds, > Answer() is usually the cause. Just as a side note... Don't forget that many other dialplan applications (Playback, Background, etc.) automatically answer the call if it hasn

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Brian J. Murrell
On Wed, 2008-06-11 at 14:51 -0400, Steve Totaro wrote: > > If you ever have problems with a call dropping after 30 seconds, > Answer() is usually the cause. Interesting. I can't say that I've ever had that problem. b. signature.asc Description: This is a digitally signed message part ___

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Steve Totaro
On Wed, Jun 11, 2008 at 2:44 PM, Brian J. Murrell <[EMAIL PROTECTED]> wrote: > On Wed, 2008-06-11 at 13:30 -0500, Brent Davidson wrote: >> On the subject of CallerID and ringing, I'm not sure if it's like this >> everywhere in the US, but where I live in Texas, our caller ID signal >> is sent betwe

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Brian J. Murrell
On Wed, 2008-06-11 at 13:30 -0500, Brent Davidson wrote: > On the subject of CallerID and ringing, I'm not sure if it's like this > everywhere in the US, but where I live in Texas, our caller ID signal > is sent between the first and second rings. It's like that here in Canada too. > If the phone

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Steve Totaro
On Wed, Jun 11, 2008 at 2:30 PM, Brent Davidson <[EMAIL PROTECTED]> wrote: > Steve Totaro wrote: > > On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro > <[EMAIL PROTECTED]> wrote: > > > On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <[EMAIL PROTECTED]> wrote: > > > On Wed, Jun 11, 2008 at 9:17 AM, Brian J.

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Brent Davidson
Steve Totaro wrote: On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <[EMAIL PROTECTED]> wrote: On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <[EMAIL PROTECTED]> wrote: I'm wondering if the SIP lines can star

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Steve Totaro
On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <[EMAIL PROTECTED]> wrote: >> On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <[EMAIL PROTECTED]> wrote: >>> I'm wondering if the SIP lines can start ringing as soon as the zap li

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Steve Totaro
On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <[EMAIL PROTECTED]> wrote: > On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <[EMAIL PROTECTED]> wrote: >> I'm wondering if the SIP lines can start ringing as soon as the zap line >> gets a call and when the zap line finally gets the CID, that is passed

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Gordon Henderson
On Wed, 11 Jun 2008, Steve Totaro wrote: > On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson > <[EMAIL PROTECTED]> wrote: >> On Wed, 11 Jun 2008, Steve Totaro wrote: >> >>> "That brings up a question though, on a regular landline with caller ID >>> the phone rings right away, it just doesn't disp

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Steve Totaro
On Wed, Jun 11, 2008 at 11:55 AM, John Novack <[EMAIL PROTECTED]> wrote: > > > Steve Totaro wrote: >> On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson >> <[EMAIL PROTECTED]> wrote: >> >>> On Wed, 11 Jun 2008, Steve Totaro wrote: >>> >>> "That brings up a question though, on a regular landlin

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread John Novack
Steve Totaro wrote: > On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson > <[EMAIL PROTECTED]> wrote: > >> On Wed, 11 Jun 2008, Steve Totaro wrote: >> >> >>> "That brings up a question though, on a regular landline with caller ID >>> the phone rings right away, it just doesn't display cal

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Raj Jain
On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <[EMAIL PROTECTED]> wrote: > I'm wondering if the SIP lines can start ringing as soon as the zap line > gets a call and when the zap line finally gets the CID, that is passed > down to the already ringing SIP phones. This is actually an interesting

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Steve Totaro
On Wed, Jun 11, 2008 at 10:57 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: > On Wed, 11 Jun 2008, Steve Totaro wrote: > >> "That brings up a question though, on a regular landline with caller ID >> the phone rings right away, it just doesn't display caller ID info >> until a couple of rings. Wh

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Brian J. Murrell
On Wed, 2008-06-11 at 15:57 +0100, Gordon Henderson wrote: > > Intersting idea... However, I live in a country where on a regular > landline with caller ID, the caller ID is displayed before the phone > rings, so make sure it's an option and not hard-wired... Well, I think your situation makes

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Gordon Henderson
On Wed, 11 Jun 2008, Steve Totaro wrote: > "That brings up a question though, on a regular landline with caller ID > the phone rings right away, it just doesn't display caller ID info > until a couple of rings. Why not have that option in Asterisk?" Intersting idea... However, I live in a countr

Re: [asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Steve Totaro
On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <[EMAIL PROTECTED]> wrote: > Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS > interface. As it is now, when the zap line gets a call, Asterisk > answers it and waits for the analog CID to be presented, then rings the > SIP phones

[asterisk-users] SIP call, updated with CID as it becomes available

2008-06-11 Thread Brian J. Murrell
Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS interface. As it is now, when the zap line gets a call, Asterisk answers it and waits for the analog CID to be presented, then rings the SIP phones with the call and the CID. There's a significant latency involved in doing this. I