[Asterisk-Users] Set context based on CID...
I was wondering if someone might be able to offer a suggestion to me about how I might go about dropping a caller into a context specific to their CID. For example, I would like to be able to dial Asterisk from a specific number (a mobile phone) and have it drop me into a context other then the one that normal callers receive that has more options tailored to things I might want to do. I assume that “answer” can somehow be used to do this but I thought I might ask the experts and see what they might have to say. Thanks in advance, (You guys are great) Matt ^ ! Matt McIntyre (KF4FGZ) ! Certified Novell Administrator ! (336) 334-1134 (Campus telephone) ! (336) 215-7199 (Mobile telephone) <- Please note the change ! (336) 334-1134 (Facsimile) ! E-MAIL: [EMAIL PROTECTED] ! AIM: MixMANJaVa ! ICQ: 11956085 ^
[Asterisk-Users] International PSTN dialing
I am interested in subscribing to a service that will let me dial the PSTN in Ireland and am interested in what the community thinks about who has the best services available. I would prefer to purchase time in blocks of minutes or pay as I go in lieu of having a monthly fee to contend with since I don’t plan to need to use it very often. Thanks for the help, Matt ^ ! Matt McIntyre (KF4FGZ) ! Certified Novell Administrator ! (336) 272-9139 (Campus telephone) ! (336) 215-7199 (Mobile telephone) <- Please note the change ! (336) 272-9139 (Facsimile) ! E-MAIL: [EMAIL PROTECTED] ! AIM: MixMANJaVa ! ICQ: 11956085 ^
[Asterisk-Users] Is IAXtel down?
For the past few days I have not been able to reach my asterisk server from the IAXtel PSTN gateway and just assumed there was a configuration error on my part. I have however not been able to find the problem and according to my asterisk console, I am not registered due to timeout. I can not make toll-free calls through IAXtel either but I can contact the demonstration server at digium with no problem. Is IAXtel still working for everyone else? Matt ^ ! Matt McIntyre (KF4FGZ) ! Certified Novell Administrator ! (336) 272-9139 (Campus telephone) ! (336) 215-7199 (Mobile telephone) <- Please note the change ! (336) 272-9139 (Facsimile) ! E-MAIL: [EMAIL PROTECTED] ! AIM: MixMANJaVa ! ICQ: 11956085 ^
RE: Spam Alert: [Asterisk-Users] Is IAXtel down?
By calling (248) 724-0700 and entering the full IAXtel number at the prompt. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Tuesday, February 24, 2004 10:58 PM To: [EMAIL PROTECTED] Subject: Re: Spam Alert: [Asterisk-Users] Is IAXtel down? Hi Matt How do you reach a Iaxtel 1-700 from PSTN? didn't think this was possible ? Barry
[Asterisk-Users] SIP and distinctive ring
Has anyone implemented distinctive ring for SIP devices in Asterisk? My searches revealed that there was a patch created at one time but I can't tell if it was accepted or not. Basically I have a Sipura analog adapter that I would like to have ring differently for "internal" calls vs external calls. Thanks guys, Matt ^^^^^ ! Matt McIntyre (KF4FGZ) ! Certified Novell Administrator ! (336) 272-9139 (Campus telephone) ! (336) 215-7199 (Mobile telephone) <- Please note the change ! (336) 272-9139 (Facsimile) ! E-MAIL: [EMAIL PROTECTED] ! AIM: MixMANJaVa ! ICQ: 11956085 ^ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?
I am experiencing this same problem and was wondering if anyone has come to a resolution. I have contacted Sipura but have not heard any response yet and am having trouble determining for sure whether the problem resides with Asterisk or the Sipura. As I have noticed that there are many users on the list who use the Sipura unit without this problem (and even a fellow with one unit that worked and one that did) I think the Sipura must be suspect. Thanks, Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, March 17, 2004 11:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura line 1 outgoing voice problem? __ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) NAT is not an issue as the Sipura and * are on the same network. Is anyone else having this problem? It looks like other people are using Sipura (I saw one user with 30 of them ?!) and am surprised that nobody else is complaining about this problem. I am willing to step through some sip debug if anyone is interested in the output. * version: Asterisk CVS-02/08/04-22:22:57 Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem would go away) Relevent config sections: --8<-- sip.conf --8<-- [cordless1] type=friend username=cordless1 secret=xxx host=dynamic context=cordless1 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw [cordless2] type=friend username=cordless2 secret=xxx host=dynamic context=cordless2 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw -- Chris ___ I had the exact same problem with a Mediatrix 1102doing a flash hook brought both sides of the conversation together. I found out that my sip.conf file had GSM as the first priority codec and the 1102 doesn't support GSM. I kept that the same but put a "disallow = gsm" statement in my sip entry for the 1102 so g.711ulaw would be the first negotiated codec. That fixed the problem. VZ
RE: [Asterisk-Users] One way audio
Upgrade your firmware on the SPA-2000 and see if it fixes the one way audio problem. I had this problem and worked with Sipura to get it resolved. If you are running a firmware earlier then version 2.0.6(c) then you will have this problem. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Mattinen Sent: Sunday, June 20, 2004 2:00 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] One way audio Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the offending line looks like this: [202] type=friend username=202 secret=voip-analog0 host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 It works fine when calling between internally, or when the SPA-2000 is the calling source, but if a call comes in on a zap channel, the one way audio problem appears. -- Seth "et lux in tenebris lucet" Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
