[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.

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 with

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 country

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 the

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. Why not have

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 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 caller ID info

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 landline with caller ID the

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 display caller ID

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 down to

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 line gets a

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 start

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. Murrell [EMAIL

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 is

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 between the

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 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

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 after

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.com

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, Digium,