[asterisk-users] Asterisk 1.6 AEL Macro vs GoSub
Hi,
I have recently run into the problem with macro implementation in AEL in
Asterisk 1.6. I have some older AEL dialplan which runs on 1.4 but it does not
on 1.6 and I'm not sure how to solve this correctly. Let me explain...
For example, in Asterisk 1.4 I have a macro like this:
macro read_digits(digits) {
Set(playlist=${SHELL(${PYTHON} ${SCRIPTS}/read_digits.py ${digits})});
if (${playlist}!=) {
Background(${playlist});
}
}
This macro calls a python script which generates a list of sound files which
are then played back by Background application. So whenever in my Dialplan I
need to read some digits, I simply do:
read_digits(20);
In 1.4 macro is implemented as macro and this is quite nice because I can use
it as follows:
context test {
s = {
read_digits(20);
}
h = {
// do something
}
}
Macro is executed in the original context and ordinary as well as special
extensions are handled by this context. As AEL is not much of a real
programming language and there aren't many possibilities how to make some parts
of code abstract, this was at least something.
But in 1.6 AEL macro has been reimplemented thru GoSub and it is translated
into context. So when the macro is performing it's work there is a need to
catch special extensions and so. The code above won't work because hangup in
read_digits macro is not catched. New macro should look like this:
macro read_digits(digits) {
Set(playlist=${SHELL(${PYTHON} ${SCRIPTS}/read_digits.py ${digits})});
if (${playlist}!=) {
Background(${playlist});
}
catch h {
// do something
}
}
But catching the h extension in the macro doesn't solve my problem as I need
to do different things in the h extension in different contexts. Only
possible workaround that comes to my mind is a copypaste of the code which
practically ruins any advantage of using a macro.
Any thoughts on how to do this in a nice way? Maybe I'm missing something...
Thanks,
Jiri Pokorny
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[asterisk-users] Monitor() - splitting long calls into several sound files
Hi, I'm not sure whether this is possible but if it is, I'm sure someone on here might know ... Is it possible to use Monitor() to record a conversation[1], but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on for a long time? We already have this happening if the callers press a specific key sequence (which we've defined in features.conf) to pause/resume recordings but I'd also like to do this automatically during long calls so that we can split the recordings up into several 'legs'. The reason for this is that the wav files spool to a ram disk[2] and if there are quite a few very long calls they can fill the ram disk up. If we could split long calls into a series of smaller files, we could move files off the ramdisk once they're no longer being actively written to and recombine them later once the call has finished. Any ideas? [1] we used to use MixMonitor() but we stopped using that for a valid reason though I can't remember what the reason was now. [2] if spooling to disk we get audio dropouts when a lot of calls are being simultaneously recorded Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE122
I have had a couple thunderstorms take out a card, again last night. The card with dahdi show status still report OK both times. When calling into the card I get all circuits are busy. However, simply replacing the card did the trick. Before that I stopped asterisk, restarted DAHDI, rebooted all those... The card always reported OK with show status. Only after replacing the card did it start to work again. I am running 1.4.42 asterisk, 2.4.1.2 DAHDI and libpri 1.4.12. So I am surprised that asterisk was not reporting the status as something other that OK - also that 2 cards have gone bad seemingly related to thunderstorms. Any thoughts? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE122
Jerry Geis wrote: I have had a couple thunderstorms take out a card, again last night. The card with dahdi show status still report OK both times. When calling into the card I get all circuits are busy. However, simply replacing the card did the trick. Before that I stopped asterisk, restarted DAHDI, rebooted all those... The card always reported OK with show status. Only after replacing the card did it start to work again. I am running 1.4.42 asterisk, 2.4.1.2 DAHDI and libpri 1.4.12. So I am surprised that asterisk was not reporting the status as something other that OK - also that 2 cards have gone bad seemingly related to thunderstorms. Any thoughts? Jerry Clearly you need to provide better protection on your incoming circuit(s) Keep in mind that the telco protection is to protect their equipment, not yours This is the case regardless of the circuit provided. John Novack -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1869 / Virus Database: 2092/4614 - Release Date: 11/13/11 -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE122
In article 4ec120c1.8080...@pagestation.com, Jerry Geis ge...@pagestation.com wrote: I have had a couple thunderstorms take out a card, again last night. The card with dahdi show status still report OK both times. When calling into the card I get all circuits are busy. However, simply replacing the card did the trick. Before that I stopped asterisk, restarted DAHDI, rebooted all those... The card always reported OK with show status. Only after replacing the card did it start to work again. Did you actually power off and on again with the original card in place? I have found in the past that a card can get in a state that just a reset is insufficient to clear, but that a power cycle does make it work again. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unavailable state not reported to Cisco SPA50X phone
Am 14.11.11 06:54, schrieb Linux: I tried to understand the rfc4235 which states the following: However, using this package to model state for non- session dialog usages is out of the scope of this specification. Does this actually mean that the device state of being offline is not part of this standard and as such can not be used to reflect the unavailable state in BLF? Hello Hans, Thats exactly the problem you hit. Even the RFC or the phones support something like an offline state for not available phones. In asterisk 1.2 this worked but it was changed after this version to be RFC conform. I have solved this problem by using a proxy in front which checkes if a phone is registered and set back a 404 if its not online (registered) but this is not a solution for you. Maybe you can build some solution with an external script and setting custom device states. best regards Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unavailable state not reported to Cisco SPA50X phone
Hello Stefan, Thank you for your answer. I was already afraid it is not in the standard. I will look into the custom device states. thanks, Hans -Original message- To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; From:Stefan Schmidt s...@sil.at Sent:Mon 14-11-2011 15:42 Subject:Re: [asterisk-users] unavailable state not reported to Cisco SPA50X phone Am 14.11.11 06:54, schrieb Linux: I tried to understand the rfc4235 which states the following: However, using this package to model state for non- session dialog usages is out of the scope of this specification. Does this actually mean that the device state of being offline is not part of this standard and as such can not be used to reflect the unavailable state in BLF? Hello Hans, Thats exactly the problem you hit. Even the RFC or the phones support something like an offline state for not available phones. In asterisk 1.2 this worked but it was changed after this version to be RFC conform. I have solved this problem by using a proxy in front which checkes if a phone is registered and set back a 404 if its not online (registered) but this is not a solution for you. Maybe you can build some solution with an external script and setting custom device states. best regards Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Application / hangup with option h or H / featuremap / more than 1 valid key
Hi, I am using the Dial-Application and need to hangup (caller and callee) with more than one key. Is it possible to set this feature, so that the caller or callee can hangup by pressing any of the keys 1, 2 or 3? I tried to configure it in the context [featuremap] in the file features.conf (yes I also did a "features reload") but with no success. None of this worked for me: [featuremap] disconnect = 123 [featuremap] disconnect = 1,2,3 I also tried to pass it to the dial application with no success: exten = 123,1,dial(number,time,h(123)H(123)) Any idea? Or is it not possible? I am using Asterisk 1.6.1.20 an have no problem with the "normal hangup option" via pressing "*". Thanks, -Thorsten- -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor() - splitting long calls into several sound files
Once the call is completed you can use SOX to split the call. In my opinion, you will have to get a larger ram disk or record the files to a different format like WAV49, but maybe somebody has a better solution for you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart Sent: Monday, November 14, 2011 7:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Monitor() - splitting long calls into several sound files Hi, I'm not sure whether this is possible but if it is, I'm sure someone on here might know ... Is it possible to use Monitor() to record a conversation[1], but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on for a long time? We already have this happening if the callers press a specific key sequence (which we've defined in features.conf) to pause/resume recordings but I'd also like to do this automatically during long calls so that we can split the recordings up into several 'legs'. The reason for this is that the wav files spool to a ram disk[2] and if there are quite a few very long calls they can fill the ram disk up. If we could split long calls into a series of smaller files, we could move files off the ramdisk once they're no longer being actively written to and recombine them later once the call has finished. Any ideas? [1] we used to use MixMonitor() but we stopped using that for a valid reason though I can't remember what the reason was now. [2] if spooling to disk we get audio dropouts when a lot of calls are being simultaneously recorded Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor() - splitting long calls into several sound files
Hi. Yeah, sox and soxmix are no problem - we're already using that to merge/join all of the segments together if people pause then resume the recordings mid call. The main issue is getting Asterisk to split the recordings into segments even when users don't pause/resume the recordings (which most don't do). The problem with ramdisks is that they're inherently limited in their storage compared to hard disks, so increasing the size of the ramdisk or changing the codec might improve matters but it's still several orders of magnitude inferior (in terms of storage). If we could get Asterisk to split the recording into segments during a call, it would allow us to raise the bar considerably. I'm doubting whether it's actually possible, but I'm hoping to be pleasantly surprised :) Cheers, Kingsley. On Mon, 2011-11-14 at 09:41 -0600, Danny Nicholas wrote: Once the call is completed you can use SOX to split the call. In my opinion, you will have to get a larger ram disk or record the files to a different format like WAV49, but maybe somebody has a better solution for you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart Sent: Monday, November 14, 2011 7:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Monitor() - splitting long calls into several sound files Hi, I'm not sure whether this is possible but if it is, I'm sure someone on here might know ... Is it possible to use Monitor() to record a conversation[1], but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on for a long time? We already have this happening if the callers press a specific key sequence (which we've defined in features.conf) to pause/resume recordings but I'd also like to do this automatically during long calls so that we can split the recordings up into several 'legs'. The reason for this is that the wav files spool to a ram disk[2] and if there are quite a few very long calls they can fill the ram disk up. If we could split long calls into a series of smaller files, we could move files off the ramdisk once they're no longer being actively written to and recombine them later once the call has finished. Any ideas? [1] we used to use MixMonitor() but we stopped using that for a valid reason though I can't remember what the reason was now. [2] if spooling to disk we get audio dropouts when a lot of calls are being simultaneously recorded Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Becoming a CLEC
I know this questions is not really asterisk related, however I know a lot of people here are in the industry. I was curious if anyone here could provide insight on how to become a facilities based CLEC. I did a lot of google-ing and read info on voip-info.org but it's all the same generic stuff and they tell you how to contact an agency. If anyone has some insider knowledge or advice, please shoot me an email. Currently, we are in the Westchester County and will be renting rack space in 60 Hudson down in NYC. We are already are an ISP/ITSP but we are a reseller and a paper CLEC. Thanks, --E -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
It is my understanding a facilities based CLEC is a FCC designation. There are rules that govern who is considered a CLEC. We are a VoIP based interconnect carrier based on the rules. Even though we offer internet services. We do not own any phone service exchanges. If you can operate out side of the CLEC classification you are much farther ahead. The amount of reporting and fees a VoIP based interconnect carrier is subject to is quite a bit less than a CLEC. Check out the FCC's website. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: eherr email.eherr9...@gmail.com Sent: Monday, November 14, 2011 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Becoming a CLEC I know this questions is not really asterisk related, however I know a lot of people here are in the industry. I was curious if anyone here could provide insight on how to become a facilities based CLEC. I did a lot of google-ing and read info on voip-info.org but it's all the same generic stuff and they tell you how to contact an agency. If anyone has some insider knowledge or advice, please shoot me an email. Currently, we are in the Westchester County and will be renting rack space in 60 Hudson down in NYC. We are already are an ISP/ITSP but we are a reseller and a paper CLEC. Thanks, --E -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do extensions stay registered
I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk I'm still here, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former. But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.comhttp://www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
SIP normally doesn't use TCP, it uses UDP, and is sessionless in that context. The exact mechanics of a registration can get deeply involved, so I'm going to give a very cursory overview. The endpoint tells the server (Asterisk, or whatever) that it would like to register, with a username and password, and what its IP address and port are. The server puts this in a list, and when it has a call for that endpoint, sends UDP packets to the known IP and port. There it typically encounters a NAT rounter, which had opened the port during the original registration and hopefully still has it open. You can enable a feature called NAT keep-alive on most endpoints to overcome bad NAT in some routers. On Mon, Nov 14, 2011 at 2:51 PM, Douglas Mortensen d...@impalanetworks.comwrote: I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? ** ** Do the extensions simply register repeatedly as a means of telling asterisk “I’m still here”, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former.* *** ** ** But am I oversimplifying it? Is there more to the process? ** ** Thanks, - Doug Mortensen Network Consultant *Impala Networks Inc* CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
Extensions do not register - peers do. A peer can register itself or be registered by Asterisk. In most cases the extension is equivalent to the peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Monday, November 14, 2011 3:52 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk I'm still here, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former. But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is UNKNOWN If I am not mistaken. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 14, 2011 5:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered Extensions do not register - peers do. A peer can register itself or be registered by Asterisk. In most cases the extension is equivalent to the peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Monday, November 14, 2011 3:52 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk I'm still here, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former. But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
Just trying to offer a little enlightenment - There are basically two methods of sip phone (peer/extension) registration. Method 1 is self-registration where Asterisk does not know or care about the phone until it asks to register. Method 2 is required-registration where Asterisk expects the phone to be there pretty much 24/7 and will attempt to register the phone and verify that it is still there at whatever frequency is specified. I personally record method 1 phones in users.conf and method 2 phones in sip.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Monday, November 14, 2011 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is UNKNOWN If I am not mistaken. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 14, 2011 5:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered Extensions do not register - peers do. A peer can register itself or be registered by Asterisk. In most cases the extension is equivalent to the peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Monday, November 14, 2011 3:52 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk I'm still here, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former. But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
The SIP server has no way to tell the device is no longer available until the next time the device registers (or the server tries to send a call to the device). ASTERISK has the qualify feature, which uses a SIP OPTIONS packet to probe the peer every min or so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Monday, November 14, 2011 5:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is UNKNOWN If I am not mistaken. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 14, 2011 5:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered Extensions do not register - peers do. A peer can register itself or be registered by Asterisk. In most cases the extension is equivalent to the peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Monday, November 14, 2011 3:52 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk I'm still here, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former. But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
I think the registration part was answered. The de-registration part is different. If the phone is gracefully taken off line it specifically de-registers. If it just can't be reached because it powers off or the router closes NAT, or whatever, then Asterisk won't know this until it times out. On Mon, Nov 14, 2011 at 3:19 PM, eherr email.eherr9...@gmail.com wrote: I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is “UNKNOWN” ** ** If I am not mistaken. ** ** --E ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Monday, November 14, 2011 5:01 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] How do extensions stay registered ** ** “Extensions” do not register – peers do. A peer can register itself or be registered by Asterisk. In most cases the “extension” is equivalent to the “peer” (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Douglas Mortensen *Sent:* Monday, November 14, 2011 3:52 PM *To:* 'asterisk-users@lists.digium.com' *Subject:* [asterisk-users] How do extensions stay registered ** ** I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? ** ** Do the extensions simply register repeatedly as a means of telling asterisk “I’m still here”, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former.* *** ** ** But am I oversimplifying it? Is there more to the process? ** ** Thanks, - Doug Mortensen Network Consultant *Impala Networks Inc* CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
I think the wrap up answer is the interval of registration compacted, if used, with the SIP OPTION packet. I like the SIP OPTION packet because we have scripts to monitor the status and lets us know when a phone is up or down. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Monday, November 14, 2011 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do extensions stay registered I think the registration part was answered. The de-registration part is different. If the phone is gracefully taken off line it specifically de-registers. If it just can't be reached because it powers off or the router closes NAT, or whatever, then Asterisk won't know this until it times out. On Mon, Nov 14, 2011 at 3:19 PM, eherr email.eherr9...@gmail.com wrote: I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is UNKNOWN If I am not mistaken. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 14, 2011 5:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How do extensions stay registered Extensions do not register - peers do. A peer can register itself or be registered by Asterisk. In most cases the extension is equivalent to the peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Monday, November 14, 2011 3:52 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk I'm still here, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former. But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 tel:%28505%29%20327-7300 F: (505) 327-7545 tel:%28505%29%20327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging Specific Verbose Level To Seperate File
Un-top-posting and de-crufting... On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer wrote: I'm using DumpChan(1001)... I would like to dump this output to a file specifically for DumpChan... On Mon, 14 Nov 2011, Warren Selby wrote: If you call DumpChan from an AGI you should be able to read the response programmatically and then dump the data into a database. Cleans up your dialplan but requires some scripting or programming knowledge (php, perl, bash or even C) in order to write the AGI. If you execute the dumpchan() application from an AGI, the output is still via Asterisk's logging mechanism. If your logging level is high enough, that output will be logged. If you log to syslogd, it will be as a single line. I don't know what it looks like if you log to files. In any case, I'd say this is a bad approach because the 'verbosity' level is too volatile. Some people (me), automagically* bump up verbosity when they access the Asterisk console and then automagically set it back to 0 when they exit the console. Perhaps executing dumpchan() via AMI would be more promising. *) Via shell scripting. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trouble with sip connection and registration
I have a home asterisk box which connects to the office asterisk, so I can just dial extensions. This used to work just fine. I'm using 10.0-rc1 on the home box, 1.8.7.0 on the office. But it doesn't work now: [Nov 14 18:38:19] NOTICE[21563]: chan_sip.c:13161 sip_reg_timeout:-- Registration for 'sip@officePBX' timed out, trying again (Attempt #86) I first thought it was some fall out of the new upgrade to 10.0-rc1, but now I'm not sure. Even with verbose at 6 I don't see anything on the office console about the attempted registration. And on the office: lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME asterisk 2045 asterisk 15u IPv4 23030 0t0 UDP *:sip but: telnet localhost 5060 Trying 127.0.0.1... telnet: connect to address 127.0.0.1: Connection refused iptables is set to allow 5060 udp and tcp. And I've flushed iptables, but still no luck. I can ssh into the office box from the home box. The office box is directly connected, the home nat'ed. Any help really appreciated. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble with sip connection and registration
You're not going to get a telnet connection on port 5060, since that's tcp and sip uses UDP. Use tcpdump/wireshark on your office pbx to see if the packets are getting to you. If not, then there's something wrong inbetween. A firewall misconfig, perhaps. Or the unthinkable: your home ISP has started filtering 5060. On Nov 14, 2011, at 18:51, sean darcy seandar...@gmail.com wrote: I have a home asterisk box which connects to the office asterisk, so I can just dial extensions. This used to work just fine. I'm using 10.0-rc1 on the home box, 1.8.7.0 on the office. But it doesn't work now: [Nov 14 18:38:19] NOTICE[21563]: chan_sip.c:13161 sip_reg_timeout:-- Registration for 'sip@officePBX' timed out, trying again (Attempt #86) I first thought it was some fall out of the new upgrade to 10.0-rc1, but now I'm not sure. Even with verbose at 6 I don't see anything on the office console about the attempted registration. And on the office: lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME asterisk 2045 asterisk 15u IPv4 23030 0t0 UDP *:sip but: telnet localhost 5060 Trying 127.0.0.1... telnet: connect to address 127.0.0.1: Connection refused iptables is set to allow 5060 udp and tcp. And I've flushed iptables, but still no luck. I can ssh into the office box from the home box. The office box is directly connected, the home nat'ed. Any help really appreciated. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble with sip connection and registration
On Mon, 14 Nov 2011, sean darcy wrote:
telnet localhost 5060
Trying 127.0.0.1...
telnet: connect to address 127.0.0.1: Connection refused
Telnet is TCP while SIP is usually UDP.
The 'Connection refused' just means you don't have telnetd running (a good
thing) or anything else (xinetd, honeypot, etc.) listening on that TCP
port.
iptables is set to allow 5060 udp and tcp...
127.0.0.1 is assigned to the 'lo' ('loopback') interface and probably not
filtered by your iptables rules.
If you enable SIP debugging on your home and office Asterisk servers, does
that yield any clues?
tshark and wireshark may also yield clues.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] Becoming a CLEC
I think that you actually should be looking to your state. I'm pretty sure that even if CLEC is an FCC designation, it is implemented either on a per-state or per-LATA basis. Here in NM there's only 1 LATA, which is why I'm not completely sure. But I believe that the CLEC qualifications designation is actually managed by the State of NM where I am. One of the state departments had all of the info seemed to be in charge. And even if there is more paperwork, reporting, red tape, etc. there are also some MAJOR discounts to be had on circuits due to the regulation that is placed on the ILECs to foster competition. They hate it. But it's not going to change any time in the foreseeable future. I looked at it back in 2006 or 2007, and there are some competencies that have to be demonstrated in both data and voice. Anyway, that's about all I know on the topic. Good luck! - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Bryant Zimmerman [mailto:brya...@zktech.com] Sent: Monday, November 14, 2011 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a CLEC It is my understanding a facilities based CLEC is a FCC designation. There are rules that govern who is considered a CLEC. We are a VoIP based interconnect carrier based on the rules. Even though we offer internet services. We do not own any phone service exchanges. If you can operate out side of the CLEC classification you are much farther ahead. The amount of reporting and fees a VoIP based interconnect carrier is subject to is quite a bit less than a CLEC. Check out the FCC's website. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: eherr email.eherr9...@gmail.commailto:email.eherr9...@gmail.com Sent: Monday, November 14, 2011 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Becoming a CLEC I know this questions is not really asterisk related, however I know a lot of people here are in the industry. I was curious if anyone here could provide insight on how to become a facilities based CLEC. I did a lot of google-ing and read info on voip-info.org but it's all the same generic stuff and they tell you how to contact an agency. If anyone has some insider knowledge or advice, please shoot me an email. Currently, we are in the Westchester County and will be renting rack space in 60 Hudson down in NYC. We are already are an ISP/ITSP but we are a reseller and a paper CLEC. Thanks, --E -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/14/2011 08:33 PM, Alex Balashov wrote: There is no free lunch. There is no such thing as an easy-peasy regulatory reclassification that gets you the same stuff you were paying before, but more cheaply. *paying for before -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Worst reason to become a CLEC: improved cost structure. Or, to be precise, it is a counterfactual reason, because it does not result in improved cost structure. This idea is driven by an incomplete understanding of what being a CLEC entails, or, for the less critically thoughtful, the free lunch fallacy. There is no free lunch. There is no such thing as an easy-peasy regulatory reclassification that gets you the same stuff you were paying before, but more cheaply. Becoming a CLEC is a totally different business model than the one you're in, and it entails magnitudinally more technological and regulatory complexity. It's really almost a different vertical. You should become a CLEC only if you want to become a CLEC, not if you want to be an ITSP with a lower cost basis, because you won't be. It is a very capital-intensive, non-trivial endeavour with high barriers to entry for a good reason. There will be people out there who will tell you that those barriers are low; they are on the bridge of failing CLECs, treading water. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Agreed. And facilities based CLEC even scarier. Regulatory / billing / PUC legals etc ugh Sent from my iPhone 4S On Nov 14, 2011, at 8:33 PM, Alex Balashov abalas...@evaristesys.com wrote: Worst reason to become a CLEC: improved cost structure. Or, to be precise, it is a counterfactual reason, because it does not result in improved cost structure. This idea is driven by an incomplete understanding of what being a CLEC entails, or, for the less critically thoughtful, the free lunch fallacy. There is no free lunch. There is no such thing as an easy-peasy regulatory reclassification that gets you the same stuff you were paying before, but more cheaply. Becoming a CLEC is a totally different business model than the one you're in, and it entails magnitudinally more technological and regulatory complexity. It's really almost a different vertical. You should become a CLEC only if you want to become a CLEC, not if you want to be an ITSP with a lower cost basis, because you won't be. It is a very capital-intensive, non-trivial endeavour with high barriers to entry for a good reason. There will be people out there who will tell you that those barriers are low; they are on the bridge of failing CLECs, treading water. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk bin file may change when running
Hi all， Recently，I met a very strange phenomenon。I found that my asterisk bin file had changed when running。I checked a lot of machines ， and the result is almost all of the bin files have taken place。 the following is the result of the calculation。 [root@callcenter beijin]# ls -l /usr/sbin/asterisk -rwxr-xr-x 1 root root 10883387 Oct 10 17:45 /usr/sbin/asterisk [root@callcenter asterisk-1.4.38]# ll main/asterisk -rwxr-xr-x 1 root root 10877291 Oct 10 17:45 main/asterisk [root@callcenter asterisk-1.4.38]# md5sum main/asterisk 1e1456aee0f094f5437da0e713755a4e main/asterisk [root@callcenter asterisk-1.4.38]# md5sum /usr/sbin/asterisk f620b91b486c665425be5175fdfb3810 /usr/sbin/asterisk Thanks in advance! -- Best regards! jordan pan Location:Shenzhen China Company:www.justcall.cn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/14/2011 07:56 PM, Douglas Mortensen wrote: I think that you actually should be looking to your state. I’m pretty sure that even if CLEC is an FCC designation, it is implemented either on a per-state or per-LATA basis. Here in NM there’s only 1 LATA, which is why I’m not completely sure. But I believe that the CLEC qualifications designation is actually managed by the State of NM where I am. One of the state departments had all of the info seemed to be in charge. CLECs are certified on the state level, by state public utility regulators, in most states known as the state PUC (public utilities commission). Being creatures of the local loop, interconnection with the ILEC is something that takes place separately in every LATA, often on somewhat different terms even within the same state. Negotiating a viable ICA (interconnection agreement) with the ILEC is one of the most important elements of success or failure, and is a massive endeavour of both personal scholarship and legal expenditure. The details of the agreement - most opt-in agreements are hundreds of pages long - are ones by which CLECs live or die, especially if they are doing a lot of local access, intra-LATA origination, or UNE facilities. And even if there is more paperwork, reporting, red tape, etc. there are also some MAJOR discounts to be had on circuits due to the regulation that is placed on the ILECs to foster competition. They hate it. But it’s not going to change any time in the foreseeable future. Yeah, discounts are nice. UNE DS1s in LATA 438 are $44/mo. Many people lick their chops at such a prospect. What these prices don't take into account is the up-front and recurring cost of: - CO backhaul (usually dark fiber of your own, sometimes ILEC fiber). - CO colocation - expensive, requires third-party vendors, and plenty of insurance. - CFA (circuit facilities assignment) - your cross-connects for UNE handoff in the CO. - EELs for dragging circuits out of COs in which you aren't colocated; you won't go into all of them, it's expensive. - COs where UNE pricing discipline is suspended because of the ILEC's finding of sufficient competition, in favour of special access. Amortise the up-front and recurring monthly cost of all those pain points and see what your new discounted rates are. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/14/2011 08:36 PM, Robert-IPhone wrote: Agreed. And facilities based CLEC even scarier. I'm curious what sort of thing would be considered a non-facilities based CLEC, since UNE-P was cancelled in 2003. There are some non-interconnected CLECs out there that exist for the sole purpose of leveraging rights of way and stuff like that, but there's not too many things you can do switchless, muxless, DACS-less and not interconnected these days. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Wow so I left before the end of resale Verizon UNE then. We ran Lucent 5E and Nortel DMS and provided facilities voice and DSL. Having a large SONET fibre infrastructure helped too. Sent from my iPhone 4S On Nov 14, 2011, at 8:53 PM, Alex Balashov abalas...@evaristesys.com wrote: On 11/14/2011 08:36 PM, Robert-IPhone wrote: Agreed. And facilities based CLEC even scarier. I'm curious what sort of thing would be considered a non-facilities based CLEC, since UNE-P was cancelled in 2003. There are some non-interconnected CLECs out there that exist for the sole purpose of leveraging rights of way and stuff like that, but there's not too many things you can do switchless, muxless, DACS-less and not interconnected these days. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
UNE is alive and well. UNE-P is what's gone. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:00 PM, Robert-IPhone rhuddles...@gmail.com wrote: Wow so I left before the end of resale Verizon UNE then. We ran Lucent 5E and Nortel DMS and provided facilities voice and DSL. Having a large SONET fibre infrastructure helped too. Sent from my iPhone 4S On Nov 14, 2011, at 8:53 PM, Alex Balashov abalas...@evaristesys.com wrote: On 11/14/2011 08:36 PM, Robert-IPhone wrote: Agreed. And facilities based CLEC even scarier. I'm curious what sort of thing would be considered a non-facilities based CLEC, since UNE-P was cancelled in 2003. There are some non-interconnected CLECs out there that exist for the sole purpose of leveraging rights of way and stuff like that, but there's not too many things you can do switchless, muxless, DACS-less and not interconnected these days. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Only through new, innovative applications. They will always deliver transport and dialtone cheaper than you. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote: Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk bin file may change when running
On 11/15/2011 02:45 AM, jordan pan wrote: Recently，I met a very strange phenomenon。I found that my asterisk bin file had changed when running。I checked a lot of machines ， and the result is almost all of the bin files have taken place。 [snip] Maybe prelink? See man prelink. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble with sip connection and registration
On 11/14/2011 07:05 PM, James Sharp wrote: You're not going to get a telnet connection on port 5060, since that's tcp and sip uses UDP. Use tcpdump/wireshark on your office pbx to see if the packets are getting to you. If not, then there's something wrong inbetween. A firewall misconfig, perhaps. Or the unthinkable: your home ISP has started filtering 5060. On Nov 14, 2011, at 18:51, sean darcyseandar...@gmail.com wrote: I have a home asterisk box which connects to the office asterisk, so I can just dial extensions. This used to work just fine. I'm using 10.0-rc1 on the home box, 1.8.7.0 on the office. But it doesn't work now: [Nov 14 18:38:19] NOTICE[21563]: chan_sip.c:13161 sip_reg_timeout:-- Registration for 'sip@officePBX' timed out, trying again (Attempt #86) I first thought it was some fall out of the new upgrade to 10.0-rc1, but now I'm not sure. Even with verbose at 6 I don't see anything on the office console about the attempted registration. And on the office: lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME asterisk 2045 asterisk 15u IPv4 23030 0t0 UDP *:sip but: telnet localhost 5060 Trying 127.0.0.1... telnet: connect to address 127.0.0.1: Connection refused iptables is set to allow 5060 udp and tcp. And I've flushed iptables, but still no luck. I can ssh into the office box from the home box. The office box is directly connected, the home nat'ed. Any help really appreciated. sean Unthinkable!! Used wireshark: I can see the REGISTER packets going out from the home router, but nothing from home:5060 shows up at the office. Bummer. Now I get to think about how to set up special ports between home and office. A great evening activity. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
Yeah! That is what I was thinking... Bringing Voice and Video under one umbrella, things like that... I actually come from a speech recognition and natural language processing background. Trying to build the voice network, and seeing how I can bring it all together. P.S. I started by getting acquainted with the proxies of course ;) Nick On Mon, Nov 14, 2011 at 9:42 PM, Alex Balashov abalas...@evaristesys.com wrote: Only through new, innovative applications. They will always deliver transport and dialtone cheaper than you. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote: Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble with sip connection and registration
On 11/14/2011 09:57 PM, sean darcy wrote: Unthinkable!! Used wireshark: I can see the REGISTER packets going out from the home router, but nothing from home:5060 shows up at the office. Bummer. Now I get to think about how to set up special ports between home and office. A great evening activity. I've always used a VPN to get around BS like that. A home router running dd-wrt connecting via PPTP to your server, problem solved. I'd also call your ISP and hold them accountable and ask why your traffic is being filtered. And which ISP are you using? That way we all know which ones like to play games with SIP packets. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do extensions stay registered
Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you definitely to look into SIP timers which tell how many time to resend a packet if no response is received and for how long to wait before thinking that the SIP packet got lost(network disconnected or end-point lost) so, qualify=yes a peer means to send-keep alives and have the NAT mechanism stay active, as soon as the SIP keep-alive packets reach a no-reply (max time)x(max retries) Asterisk marks the peer as UNREACHABLE. qualify=no wouldn't do all of the above. Another interesting thing to know is that SIP end-points have registrations time-out and refresh Registration timers as well. So if everything is going well, SIP end-points refresh their registration after some defined time. On Tue, Nov 15, 2011 at 3:35 AM, eherr email.eherr9...@gmail.com wrote: I think the wrap up answer is the interval of registration compacted, if used, with the SIP OPTION packet. ** ** I like the SIP OPTION packet because we have scripts to monitor the status and lets us know when a phone is up or down. ** ** --E ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Carlos Alvarez *Sent:* Monday, November 14, 2011 5:30 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How do extensions stay registered ** ** I think the registration part was answered. The de-registration part is different. If the phone is gracefully taken off line it specifically de-registers. If it just can't be reached because it powers off or the router closes NAT, or whatever, then Asterisk won't know this until it times out. ** ** On Mon, Nov 14, 2011 at 3:19 PM, eherr email.eherr9...@gmail.com wrote:* *** I think the question is more along the lines of how does asterisk know immediately when a sip phone becomes on line and when you unplug the phone from the network, how does asterisk essentially know immediately that it status is “UNKNOWN” If I am not mistaken. --E *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Monday, November 14, 2011 5:01 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] How do extensions stay registered “Extensions” do not register – peers do. A peer can register itself or be registered by Asterisk. In most cases the “extension” is equivalent to the “peer” (301 = 301) but it can be quite different (301 = sipuser1) or (301 = d...@impalanetworks.com). *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Douglas Mortensen *Sent:* Monday, November 14, 2011 3:52 PM *To:* 'asterisk-users@lists.digium.com' *Subject:* [asterisk-users] How do extensions stay registered I know this is probably a very basic question for many on this list. But in troubleshooting an issue, I wanted to take a step back ask the question. In Asterisk (or maybe all SIP), how do extensions stay registered with the SIP server? Do the extensions simply register repeatedly as a means of telling asterisk “I’m still here”, or are there actual keepalive packets that are transmitted to actually keep a TCP session alive? My guess is the former.* *** But am I oversimplifying it? Is there more to the process? Thanks, - Doug Mortensen Network Consultant *Impala Networks Inc* CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- Carlos Alvarez TelEvolve 602-889-3003 ** ** ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] Becoming a CLEC
There are clever ways to be a CLEC, and keen reasons for becoming so. But cheaper stuff ain't one of them. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 10:02 PM, Nick Khamis sym...@gmail.com wrote: Yeah! That is what I was thinking... Bringing Voice and Video under one umbrella, things like that... I actually come from a speech recognition and natural language processing background. Trying to build the voice network, and seeing how I can bring it all together. P.S. I started by getting acquainted with the proxies of course ;) Nick On Mon, Nov 14, 2011 at 9:42 PM, Alex Balashov abalas...@evaristesys.com wrote: Only through new, innovative applications. They will always deliver transport and dialtone cheaper than you. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote: Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calling an independent gateway from asterisk
Hello, I have a testing scenario at hand. I want to make a call from Asterisk CLI or AMI to an external network gateway. Is this possible. Let me explain the use case. Asterisk server (say 192.168.5.10) has few registered endpoints or softphone. Now an external gateway (say my-example.com or XXX.XXX.XXX.XXX:5060), listening for SIP invites, but this gateway is not registered with Asterisk, can I send out SIP invites (call) to this external gateway, without having to register on Asterisk. -- Thank you... Amar Akshat Please excuse any spelling mistakes, as this email was sent from a not so good mobile device. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling an independent gateway from asterisk
Hey,
Though your requirements are unclear and below may not exactly fit your
specs unless you give some more usage details.
if your gateway requires no authentication, yes you can do this by writing
a dialplan extension like below
exten = calling-togw,1,NOOP(I'll be getting some variables from AMI caller)
same = n,DIAL(SIP/${CALLTHIS}@my-example.com)
Now, in the AMI script you need to do the following.
1- Connect to asterisk,
2- Set the variable CALLTHIS as the destination you want to dial-out
3- use the
Originate-AMIhttp://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originateaction
properly.
Regards,
Sammy
On Tue, Nov 15, 2011 at 11:01 AM, Amar Akshat amar.aks...@gmail.com wrote:
Hello,
I have a testing scenario at hand. I want to make a call from Asterisk
CLI or AMI to an external network gateway. Is this possible.
Let me explain the use case.
Asterisk server (say 192.168.5.10) has few registered endpoints or
softphone.
Now an external gateway (say my-example.com or XXX.XXX.XXX.XXX:5060),
listening for SIP invites, but this gateway is not registered with
Asterisk,
can I send out SIP invites (call) to this external gateway, without
having to register on Asterisk.
--
Thank you...
Amar Akshat
Please excuse any spelling mistakes, as this email was sent from a
not so good mobile device.
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Re: [asterisk-users] Calling an independent gateway from asterisk
Amar, In general, gateways don't register. They are simply defined as a peer and calls are routed to them in the dialplan. When I do this I usually use the local channel to get to the dialing contexts. Get in touch if you need a more detailed example Bruce Ferrell On 11/14/2011 10:01 PM, Amar Akshat wrote: Hello, I have a testing scenario at hand. I want to make a call from Asterisk CLI or AMI to an external network gateway. Is this possible. Let me explain the use case. Asterisk server (say 192.168.5.10) has few registered endpoints or softphone. Now an external gateway (say my-example.com or XXX.XXX.XXX.XXX:5060), listening for SIP invites, but this gateway is not registered with Asterisk, can I send out SIP invites (call) to this external gateway, without having to register on Asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
