Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anthony Francis Sent: Tuesday, August 07, 2007 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service Douglas Garstang wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of SIP Sent: Monday, August 06, 2007 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service Steve Totaro wrote: Anthony Francis wrote: Tim Panton wrote: On 5 Aug 2007, at 06:54, Douglas Garstang wrote: I don't think creating a network without a single point of failure is unreasonable. It's impossible. I can't think of a single example where this actually exists. Getting even close is hideously expensive. Tim, speaking for himself :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In fact, the only people who would say something like this are folks who have never PHYSICALLY implemented a network, they simply don't understand the limitations involved. I worked for a CLEC in Montana, not Silicon Valley, not Manhatten, but rather PODUNK, Montana. We had redundant multi-homed servers, connected to multiple switches, running OSPF. A failure in any component (server, network, cable) would cause a failover to a backup component in about 6 seconds. We had multiple upstream providers. The servers where divided between multiple racks, split between different power plants. We did just about everything we could to make the setup redundant. The CPE equipment at any single location might fail, and that wasn't redundant, but at least if that failed, it would not affect any other customers. CPE equipment included POE enabled phones, a UPS, a POE switch and power being delivered from our plant. Yes, all the equipment was located at the same physical location. In hindsight, we could have multi-homed our collocations. Why can't service providers multi home their edge systems to accept incoming calls from two physical locations? If a service provider did this, they would have two completely independent facilities, potentially thousands of miles apart, connected to different upstream providers. I can't think of anything short of nuclear war that would destroy their ability to accept calls. If they did least cost routing, it wouldn't even matter if their providers failed. China gets hit by a meteor and NO provider can deliver calls to China? Fine... at least you can still call everywhere else. Maybe it still had some holes, but jeez, at least we tried to deliver high quality service. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There is no one here not doing best-effort redundancy, what the first gentleman had said was a network with NO single points of failure. Clearly that is a pipe dream. To the person with six second failover, that 6 seconds would have dropped calls and dialing out issues resulting in complaints. You would then tell your customer that you got it working immediately and often they don't care, they are still angry about the dropped call. MY point is, VOIP is good, great even, but anyone expecting a less than 20 year old tech to be more reliable than a tech that has been around for over a hundred (PSTN) needs to spend some more time thinking about that. So you've never gotten a dropped call or dead air on a PSTN call? Put it in a little perspective. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, August 07, 2007 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service Brian Capouch wrote: Stephen Bosch wrote: PSTN service still sets the standard. With infrastructure paid for under a gracious guaranteed-profit monopoly by ratepayers, In a regulated marketplace with legislated minimum service levels. In Canada, most of the phone systems were government-owned. It was a good system, at least from the point of view of reliability. I don't miss the surly (and often slow) service, but it's arguable whether today's service -- in which everyone smiles nice and *pretends* to serve you while ignoring you completely -- is any better. At least the bloody stuff worked. Communications infrastructure is a strategic, national asset, and only really useful if it goes everywhere, even to the unprofitable pockets like Podunk Corners, North Dakota. People forget this. In a totally free marketplace, Podunk Corners waits years for service and gets tin cans and string when it finally arrives. I disagree. There is more competition in smaller towns and rural areas. It isn't cost effective for the bigger carriers to move in, so the small ones do. They get state/federal subsidies. I'll bet you there's more ISP's, and CLEC's per square inch in Montana than there is in the bay area. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Paglayan Sent: Tuesday, August 07, 2007 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service On Aug 6, 2007, at 10:42 AM, Stephen Bosch wrote: Eric ManxPower Wieling wrote: Douglas Garstang wrote: Let's assume for a moment that it's impossible. That does not mean adding additional servers and additional networking equipment does not add value, or is a worthless endeavour. I agree with that. At least two people that I know run ITSPs. Each time they have an outage (which is not very often) they DO learn from the experience and work to avoid a future outage cause by the same issue. You would be surprised at how many little things can cause an outage. My own experience is that increasing failover redundancy, which adds correspondingly increasing complexity, also increases the odds of an outage. It is very rare that failover redundancy works as intended during an actual failover, no matter how many times you simulate it. I would rather have a simple network design where the cause of failure, when it happens, is obvious and quickly corrected. For example, I would rather have replacement parts on the shelf and be able to slap them in quickly than be running hot standbys and paying for the electricity, and then have the thing break anyway when there's a failure. I'll second that, specially for smaller installations, You must have the kind of customers that don't mind having no phone service for a few hours. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of SIP Sent: Monday, August 06, 2007 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service Steve Totaro wrote: Anthony Francis wrote: Tim Panton wrote: On 5 Aug 2007, at 06:54, Douglas Garstang wrote: I don't think creating a network without a single point of failure is unreasonable. It's impossible. I can't think of a single example where this actually exists. Getting even close is hideously expensive. Tim, speaking for himself :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In fact, the only people who would say something like this are folks who have never PHYSICALLY implemented a network, they simply don't understand the limitations involved. Anthony What if a train derails and slices through the main fiber connections. OK, so you have XO, Global Crossing, Verizon, and UCN all for redundancy. Well guess what? They are all most likely running over those strands of fiber. You better have a VSAT connection too! Good grief. No, you have two physical collocations. One in say in Nevada or Idaho (least likely states to suffer natural disasters) and one in New York. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Monday, August 06, 2007 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service Eric ManxPower Wieling wrote: Douglas Garstang wrote: Let's assume for a moment that it's impossible. That does not mean adding additional servers and additional networking equipment does not add value, or is a worthless endeavour. I agree with that. At least two people that I know run ITSPs. Each time they have an outage (which is not very often) they DO learn from the experience and work to avoid a future outage cause by the same issue. You would be surprised at how many little things can cause an outage. My own experience is that increasing failover redundancy, which adds correspondingly increasing complexity, also increases the odds of an outage. It is very rare that failover redundancy works as intended during an actual failover, no matter how many times you simulate it. I would rather have a simple network design where the cause of failure, when it happens, is obvious and quickly corrected. For example, I would rather have replacement parts on the shelf and be able to slap them in quickly than be running hot standbys and paying for the electricity, and then have the thing break anyway when there's a failure. This might work for a web service, but people have a zero tolerance for no phone service. They expect to be able to pick up their handset, and get a functional dialtone immediately. Adding additional servers, additional network components, and some smarts into your design saves being woken at 3am when a server fails. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of SIP Sent: Monday, August 06, 2007 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service Steve Totaro wrote: Anthony Francis wrote: Tim Panton wrote: On 5 Aug 2007, at 06:54, Douglas Garstang wrote: I don't think creating a network without a single point of failure is unreasonable. It's impossible. I can't think of a single example where this actually exists. Getting even close is hideously expensive. Tim, speaking for himself :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In fact, the only people who would say something like this are folks who have never PHYSICALLY implemented a network, they simply don't understand the limitations involved. I worked for a CLEC in Montana, not Silicon Valley, not Manhatten, but rather PODUNK, Montana. We had redundant multi-homed servers, connected to multiple switches, running OSPF. A failure in any component (server, network, cable) would cause a failover to a backup component in about 6 seconds. We had multiple upstream providers. The servers where divided between multiple racks, split between different power plants. We did just about everything we could to make the setup redundant. The CPE equipment at any single location might fail, and that wasn't redundant, but at least if that failed, it would not affect any other customers. CPE equipment included POE enabled phones, a UPS, a POE switch and power being delivered from our plant. Yes, all the equipment was located at the same physical location. In hindsight, we could have multi-homed our collocations. Why can't service providers multi home their edge systems to accept incoming calls from two physical locations? If a service provider did this, they would have two completely independent facilities, potentially thousands of miles apart, connected to different upstream providers. I can't think of anything short of nuclear war that would destroy their ability to accept calls. If they did least cost routing, it wouldn't even matter if their providers failed. China gets hit by a meteor and NO provider can deliver calls to China? Fine... at least you can still call everywhere else. Maybe it still had some holes, but jeez, at least we tried to deliver high quality service. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
I don't think creating a network without a single point of failure is unreasonable. -Original Message- From: [EMAIL PROTECTED] on behalf of Stephen Bosch Sent: Sat 8/4/2007 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service SIP wrote: There are also lots of big carriers masquerading as big carriers. ;) *lol* If the ONLY people who could get into the business were the ones who could, before offering any services to customers, afford to build out multiple edge systems for accepting incoming calls, each with multiple interfaces connected to multiple subnets via multiple switches using multiple upstream providers, you would have ONE single choice for an ITSP. And ATT doesn't have that amount of redundancy in their network. Working in the carrier networking business, I can assure you that we've NEVER run across a SINGLE carrier network (not from the largest to the smallest) that has redundancy in ALL aspects (or even MOST aspects) of its network. This is why there are uptime policies that allow a percentage of outages to occur. Triple 9 uptime (Exceedingly rare, but a purported goal -- 99.999%) still allows 15 full hours of downtime a year. And that rarely includes the occasional lost packet or latency. In other words, you can blame the marketing departments in various big carriers for creating these unrealistic expectations in the marketplace :) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Let's assume for a moment that it's impossible. That does not mean adding additional servers and additional networking equipment does not add value, or is a worthless endeavour. -Original Message- From: [EMAIL PROTECTED] on behalf of Tim Panton Sent: Sun 8/5/2007 5:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service On 5 Aug 2007, at 06:54, Douglas Garstang wrote: I don't think creating a network without a single point of failure is unreasonable. It's impossible. I can't think of a single example where this actually exists. Getting even close is hideously expensive. Tim, speaking for himself :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Measuring Jitter in Asterisk
How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring Jitter in Asterisk
Did a little research. Wireshark can graph jitter measurement. That's cool, but pretty useless. Now, what would be REALLY cool, was if tshark, the command line tool, could measure jitter. It looks like it lacks this feature. If it COULD, you could leave a tshark process running, constantly measuring jitter in real time. You'd run one for each ITSP you use, and voila, you have real time jitter metrics on a provider by provider basis. But... tshark doesn't' support this. Arrgh! Doug. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, August 03, 2007 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Measuring Jitter in Asterisk How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring Jitter in Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, August 03, 2007 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Measuring Jitter in Asterisk On Fri, 3 Aug 2007, Douglas Garstang wrote: If it COULD, you could leave a tshark process running, constantly measuring jitter in real time. You'd run one for each ITSP you use, and voila, you have real time jitter metrics on a provider by provider basis. There are various command-line SIP performance test tools (sipp?) that can do this too, I think. I don't think you could do this with SIPP Also, it may be possible to modify Wireshark's plugin to periodically invoke its jitter analysis function automatically and export the results to some retrievable location. The most difficult problem would be giving it a particular data stream to home in on as a VoIP call; the easiest thing there would be to nail up your own periodic tests from a SIP UAC with definable IP endpoint locations and constantly run it with that filter. Hackjobs aside, this sort of thing is essentially what products like Brix do, as well as check in with SRTP stats. Ok, maybe I should call them. But, as I said, if all their product does is measure QoS and then give you pretty graphs to eyeball, it isn't much use. I need something that can measure jitter, latency etc in real time and then stick the results somewhere, such as in MySQL. I can then choose ITSP's based not just on route cost, but on a combination of route cost and historical QoS data. Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring Jitter in Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Friday, August 03, 2007 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Measuring Jitter in Asterisk On Fri, 2007-08-03 at 12:31 -0700, Douglas Garstang wrote: How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. You can use Wireshark (formerly Ethereal) to analyze the RTP stream after it's been captured. You can either use Wireshark itself to do the network capture, or you can capture the traffic with tcpdump and then pull the file into Wireshark at a later time. Jared, that won't do. I don't want to run the wireshark GUI, and I don't wan't to run it on every single Asterisk box, connecting back to a local X server running on my desktop. I also don't want to capture the RTP data, and store it somewhere for later analysis. I'm looking at a situation here with millions of subscribers and dozens of ITSP's. What I do want to do is record QoS data to every single ITSP in real time. I can then lease cost route based not just on route cost, but also on historical QoS data. Whatever tool is used to collect the QoS data has to stick it somewhere, such as MySQL, and then when I route a call, I will have to query that data from MySQL. Inside Wireshark, go to Statistics, RTP, Show All Streams, and then select a stream and hit the Analyze button. I'm trying to avoid post-eyeballing the data. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring Jitter in Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, August 03, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Measuring Jitter in Asterisk On Fri, 3 Aug 2007, Jared Smith wrote: If the provider sends RTCP packets, you could simply watch for those and write the data to a database. (I think modern versions of Asterisk even allow you to get to the data from the dialplan, and possibly from the Manager Interface.) That at least gives you some per-call statistics. If you want to go that route, just yank those packets out of a constantly running tcpdump process with the right filters, and then process them with a script and load that data into a DB. Alex, ok... so if I wanted to measure jitter to an ITSP I could run tcpdump to it, and parse the output. According to http://wiki.wireshark.org/RTP_statistics, I'd have to compare the timestamp in each RTP packet with the timestamp shown by tcpdump. Looks kinda complicated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring Jitter in Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Todd Sent: Friday, August 03, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Measuring Jitter in Asterisk At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote: How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. You could look at the txjitter and rxjitter values (and other values) stored in the CHANNEL() function, and those values you're looking for were previously known as RTPAUDIOQOS. Or is this not sufficient? Thanks John. Missed those... they're not documented... not even in 'show function CHANNEL'. Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring Jitter in Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Todd Sent: Friday, August 03, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Measuring Jitter in Asterisk At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote: How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. You could look at the txjitter and rxjitter values (and other values) stored in the CHANNEL() function, and those values you're looking for were previously known as RTPAUDIOQOS. Or is this not sufficient? Are txjitter and rxjitter working reliably? These calls are going to be placed from AMI and bridged together. Do you think the variables would be correctly set for each leg of the call? Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ira Sent: Thursday, August 02, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Teliax Quality of Service At 09:23 AM 8/2/2007, you wrote: I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? Teliax has been quite good. I was having problems the last 2 days and they confirmed that they are working on fixing something. I've been using IP for all my outgoing calls for the last couple of years and other than being ripped off by a couple of vendors and the occasional connection problem it's saved me large amounts of money, more than what I lost when the 2 providers refused to return my deposits and then went under, but I do have ways to get dial tone on my POTS lines for those times when it all goes to heck. I confused by this. Don't ITSP's have redundancy? Don't they have multiple edge systems for accepting incoming calls? Don't their multiple edge systems have multiple interfaces, connected to multiple subnets, via multiple switches? And, don't they have multiple upstream providers? About the only thing that could go wrong that affects all service like this would be a badly pushed out software update, affecting all systems? Doug. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 320 - Can it actually be configured?
Don't know about the 320, but we provisioned the 301's. They're provisioning is basically the same as the 501's and 601's. What problems are you having? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Sent: Wednesday, August 01, 2007 2:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 320 - Can it actually be configured? Just got one of these. Horrible to program. Trying to key in the FTP server. Won't even remember the info after rebooting. Anybody know the proper way to beat on this stupid beast so it will work? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retail DID provider ?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of SIP Sent: Wednesday, August 01, 2007 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Retail DID provider ? IdeaSIP, Voxbone, Gizmo Project, etc... if we're talking retail. N. Mail list wrote: I am looking for a retail DID provider which should provide unlimited incoming calls something around 4-5 bucks . Nufone seemed like a good choice at $5 but they are down again :( . Any suggestions please . Funny how I keep seeing that. Don't any of these SIP providers have some sort of redundancy??? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different SIP From and Auth?
Looks like this isn't possible. I wonder if there's a bug open on this? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Ma Sent: Thursday, July 12, 2007 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Different SIP From and Auth? Hi I have asked this questions,but have no answer :) I also want Asterisk do not check to head with digest username in registration,how can we do that? On 7/12/07, Douglas Garstang [EMAIL PROTECTED] wrote: Is it possible to have Asterisk allow the From address in a SIP invite to be different to the required digest username? The auth parameter supposedly allows it, but whether or not I set auth to be what the UA sends as the digest username, Asterisk just complains that the from and the digest are different, and it gives up. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different SIP From and Auth?
Is it possible to have Asterisk allow the From address in a SIP invite to be different to the required digest username? The auth parameter supposedly allows it, but whether or not I set auth to be what the UA sends as the digest username, Asterisk just complains that the from and the digest are different, and it gives up. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different SIP From and Auth?
Bloody good question. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Ma Sent: Thursday, July 12, 2007 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Different SIP From and Auth? Hi I have asked this questions,but have no answer :) I also want Asterisk do not check to head with digest username in registration,how can we do that? On 7/12/07, Douglas Garstang [EMAIL PROTECTED] wrote: Is it possible to have Asterisk allow the From address in a SIP invite to be different to the required digest username? The auth parameter supposedly allows it, but whether or not I set auth to be what the UA sends as the digest username, Asterisk just complains that the from and the digest are different, and it gives up. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Early Media have to be Ulaw?
I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid=Test hone 1 +19256002182 host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default insecure=port dtmfmode=rfc2833 canreinvite=yes qualify=yes disallow=all ;allow=ulaw allow=g729 Level 3 sends early media... --- Transmitting (no NAT) to xxx.yyy.34.210:5061 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.yyy.34.210:5061;branch=z9hG4bK-tenor-d802-22d2-004d;received=xxx.yyy .34.210 From: sip:[EMAIL PROTECTED];tag=d80222d2-27 To: sip:[EMAIL PROTECTED];tag=as4fe079a5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] ontent-Type: application/sdp Content-Length: 261 v=0 o=root 2235 2235 IN IP4 xxx.yyy.34.195 s=session c=IN IP4 xxx.yyy.34.195 t=0 0 m=audio 10484 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv and Asterisk responds on the console with: [Jun 22 10:06:03] WARNING[32573]: channel.c:2882 set_format: Unable to find a codec translation path from g729 to slin [Jun 22 10:06:03] WARNING[32573]: indications.c:121 playtones_alloc: Unable to set 'SIP/19256002182-096ac918' to signed linear format (write) This doesn't happen when progressinband=no. It almost seems like Asterisk has to do early media as G711 only. Is that the case??? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug in Ex-Girlfriend logic?
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/5000-0a281f80, ) in new
stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/5000-0a281f80, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/5000-0a281f80, 19256002182)
in new stack
-- Executing [EMAIL PROTECTED]:4] Playback(SIP/5000-0a281f80,
tt-monkeys) in new stack
-- SIP/5000-0a281f80 Playing 'tt-monkeys' (language 'en')
However, when I change the extension match to:
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the
caller id number is 19256002182 from the NoOp() when it does work.
This had me stumped for a while, until I realized that the following
_DOES_ work:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,NoOp(Foo)
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
Yes. That's right. In order for the ex-girlfriend logic to match a
caller id of 19256002182 against 5000, the same context also needs to
have an extension for 5000, even if you intend to do nothing with it.
I'd never noticed this before, because normally you'd provision the 5000
extension FIRST and then the 5000/19256002182 after that.
Seems like a bug to me Problem was reproduced in 1.2.13, 1.2.19 and
1.4.4.
Doug.
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Re: [asterisk-users] 180 Ringing with SDP
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Monday, June 18, 2007 5:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 180 Ringing with SDP On Mon, 18 Jun 2007, Jared Smith wrote: I could be totally off base here, but it's my understanding that a 180 is telling Asterisk to generate ringing on it's side, and that a 183 (with SDP) would tell Asterisk that the call is progressing and that it should play the early media specified in the SDP. I'm sure someone there's probably someone on the list who is more intimate with the details of SIP that can enlighten us further on the subtle differences between the 180 and 183 provisional responses. A cursory interpretation of the RFC suggests that 180 Ringing is a message designed solely to convey ringback, and that it is the payload of the 183 response that may be used to convey additional details about the nature of the call's progress. Therefore, a 180 would be an inappropriate vehicle for early media SDP information. Tell that to level 3. :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ex-Girlfriend Logic in 1.4.4
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/5000-0a281f80, ) in new
stack
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/5000-0a281f80, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/5000-0a281f80, 19256002182)
in new stack
-- Executing [EMAIL PROTECTED]:4] Playback(SIP/5000-0a281f80,
tt-monkeys) in new stack
-- SIP/5000-0a281f80 Playing 'tt-monkeys' (language 'en')
However, when I change the extension match to:
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the
caller id number is 19256002182 from the NoOp() when it does work. What
am I missing here?
Doug.
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[asterisk-users] SIP Termination with automatic debit
Can anyone recommend any wholesale SIP termination providers that will automatically charge a credit card? Most seem to want upfront payment and a credit balance but that sucks when you have to watch it like a hawk to make sure the balance never hits zero. I'm looking for a provider that can automatically charge a credit card. Douglas. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 180 Ringing with SDP
We're dialing a disconnected number via Level 3's vector network, and are receiving this. The response has SDP in it. Apparently, Level 3 is playing early media. Asterisk doesn't seem to know what to do with SDP in a 180 RINGING, and just plays ringing. What am I missing here? How can Asterisk see there's SDP, early media, in the response and act accordingly? SIP/2.0 180 Ringing. Via: SIP/2.0/UDP xxx.yyy.34.195:5060;branch=z9hG4bK591743a1;rport=5060. From: Test Phone 2 sip:[EMAIL PROTECTED];tag=as7e76044e. To: sip:[EMAIL PROTECTED];tag=gK0cc2d5ab. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Contact: sip:[EMAIL PROTECTED]:5060. Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS. Content-Length: 231. Content-Disposition: session; handling=required. Content-Type: application/sdp. . v=0. o=Sonus_UAC 22562 17424 IN IP4 4.55.16.99. s=SIP Media Capabilities. c=IN IP4 4.55.16.66. t=0 0. m=audio 6288 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=maxptime:20. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Reload in 1.4 clears regexten
Brad, I can't post the entire contents of sip.conf and extensions.conf/extensions.ael, but as you can see below, I don't have a sip_autoreg defined anywhere in my dial plan. [EMAIL PROTECTED] asterisk]# cat sip.conf [general] context=default allowoverlap=no bindport=5060 bindaddr=xxx.yyy.34.201 srvlookup=yes regcontext=sip_autoreg [EMAIL PROTECTED] asterisk]# grep sip_autoreg extensions.conf [EMAIL PROTECTED] asterisk]# grep sip_autoreg extensions.ael [EMAIL PROTECTED] asterisk]# Douglas. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, Bradley Sent: Thursday, June 07, 2007 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Reload in 1.4 clears regexten Please post the relevant portions of your sip.conf and extensions.conf I'll bet dollars to donuts you have the same context defined as both your regcontext and as a context in extensions.conf (or an .ael, or whatever). - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, June 06, 2007 7:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Reload in 1.4 clears regexten Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi and reinvites...
I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there's only one Asterisk box between Phone1 and Phone2? Like this... +---+ +---+ | A | | B | /+---+\+---+ / \ Phone1 ---Phone2 Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Provisioning Linksys PAP2T ATA's
Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provisioning Linksys PAP2T ATA's
How do you get PAP2T-NA's? They aren't even on Linksys's web site. -Original Message- From: Doug [mailto:[EMAIL PROTECTED] Sent: Thursday, June 07, 2007 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Douglas Garstang Subject: Re: [asterisk-users] Provisioning Linksys PAP2T ATA's At 11:44 6/7/2007, Douglas Garstang, wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C7A923.2703ACD7 Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. Don't know, but would like to find out. By the way, the T in PAP2T stands for Trash. We've had about a 70% failure rate. Get the PAP2-NA's with the blue LEDs if you can. Green LEDs = Grief. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Provisioning Linksys PAP2T ATA's
Steve, Thanks, but, we can't use tftp. The ATA's (all 1500 of them) are on remote networks. As far as I know, tftp only works across a local subnet. I called Linksys and they told me the ATA's can be provisioned with http/https, but only after we become a certified reseller/provider. Gonna have to work on that I guess. Doug. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, June 07, 2007 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provisioning Linksys PAP2T ATA's On Thu, 7 Jun 2007, Douglas Garstang wrote: Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. If it's like the pap2, you can use tftp and xml. This should get you started. /tftpboot/spa000F66A83C90.xml: ?xml version=1.0 encoding=ISO-8859-1? flat-profile !-- tag case appears to be important -- !-- system, system configuration -- Admin_Passwd/Admin_Passwd User_Passwd/User_Passwd !-- system, optional network configuration -- HostNameexample/HostName Domainexample.com/Domain Primary_DNS192.168.0.4/Primary_DNS Secondary_DNS192.168.0.4/Secondary_DNS DNS_Server_OrderDHCP,Manual/DNS_Server_Order Syslog_Server192.168.0.4/Syslog_Server Debug_Server192.168.0.4/Debug_Server !-- provisioning, configuration profile -- Profile_Rule_B[--key $K] tftp://tftp.example.com:$P/spa000F66A83C90.xml/Profile_Rule_B !-- line 1, proxy and registration -- Proxy_1_dt/Proxy_1_ !-- line 1, subscriber information -- Display_Name_1_example-line-1/Display_Name_1_ User_ID_1_example-line-1/User_ID_1_ Password_1_example-line-1/Password_1_ !-- line 1, proxy and registration -- Proxy_2_sip.example.com/Proxy_2_ !-- line 1, subscriber information -- Display_Name_2_example-line-2/Display_Name_2_ User_ID_2_example-line-2/User_ID_2_ Password_2_example-line-2/Password_2_ /flat-profile Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DUNDi and reinvites...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Thursday, June 07, 2007 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi and reinvites... On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: Correct so far... although once the call is made, it's no longer a DUNDi question, and is simply a signalling question. (In other words, DUNDi is used for Phone 1 to figure out how to connect to Phone 2, but after it's figured that out, it's a normal SIP or IAX call between Asterisk A and Asterisk B.) Hi Jared. Understood. Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there's only one Asterisk box between Phone1 and Phone2? Like this... Yes, as long as the protocols are all the same. If Phone1 is talking SIP to Asterisk A and Asterisk A is talking IAX to Asterisk B and Asterisk B is talking SIP to Phone 2, then it won't happen. But assuming everything is using the same transport, they'll happen. In fact, if re-invites are enabled on both Asterisk servers, and the two phones can communicate directly, you can re-invite *both* Asterisk servers out of the middle of the call. I figured the protocols would have to be the same. The phones are SIP based, so I tried to get DUNDi to work with SIP. That's where I hit snags. The INVITE coming from Asterisk 1 has the original phone's From: address, because it's much easier for Asterisk 2 to accept calls from Asterisk 1 based on the IP address. However, because the INVITE still has the original phones FROM: tag, Asterisk matches it against it's own copy of Phone 1's sip entry, rather than the entry for Asterisk 1, and then sends a 407 proxy Auth message back to the Asterisk 1, who doesn't know what to with it. Another, much uglier approach, is to change the From Address that Asterisk 1 sends the INVITE with. However, then we'd need to add extra SIP headers to the INVITE going out from Asterisk 1. Asterisk two would authenticate against those and pluck out the extra SIP headers to get the original caller id. I also tried setting the username/secret on Asterisk 1 for it's trunk to Asterisk 2, thinking that the From: and the auth credentials would be different, but Asterisk threw a fit when the From: did not match the digest id. What's wrong with that? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DUNDi and reinvites...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Thursday, June 07, 2007 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi and reinvites... On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: Correct so far... although once the call is made, it's no longer a DUNDi question, and is simply a signalling question. (In other words, DUNDi is used for Phone 1 to figure out how to connect to Phone 2, but after it's figured that out, it's a normal SIP or IAX call between Asterisk A and Asterisk B.) Is there a way configure re-invites in this situation so that either Asterisk A or B drops out of the call, and there's only one Asterisk box between Phone1 and Phone2? Like this... Yes, as long as the protocols are all the same. If Phone1 is talking SIP to Asterisk A and Asterisk A is talking IAX to Asterisk B and Asterisk B is talking SIP to Phone 2, then it won't happen. But assuming everything is using the same transport, they'll happen. In fact, if re-invites are enabled on both Asterisk servers, and the two phones can communicate directly, you can re-invite *both* Asterisk servers out of the middle of the call. Jared, we also don't want to reinvite all the way down to the two phones communicating with each other. We want a single Asterisk system between them. I just reconfigured my setup to send calls from Asterisk 1 to Asterisk 2 with a callerid/From: different to the originating phone's, just to get to the point where I can set reinvites up. Let's just say we only configured the originating phone with canreinvite=yes, which hopefully means the originating phone would reinvite with the second Asterisk server. That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DUNDi and reinvites...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, June 07, 2007 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DUNDi and reinvites... Douglas Garstang wrote: Let's just say we only configured the originating phone with canreinvite=yes, which hopefully means the originating phone would reinvite with the second Asterisk server. That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! reinvites only remove the server from the AUDIO PATH. Signaling is still going thru Asterisk no matter what happens. *nod* I know. :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reload in 1.4 clears regexten
Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will clear any extensions that have been created by regexten. This is VERY bad! Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 Includes in Macro...
Where's Steve Murphy when you need him? :-)
This doesn't seem to work in AEL2...
Macro foo(arg1) {
.
Includes {
Hangup;
}
}
The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
Error: syntax error, unexpected KW_INCLUDES, expecting '}'
The same error does not occur when the includes is in a context.
I need to have the ability to include my hangup routine in macros, as
theoretically, a hang up could occur while asterisk is processing code
from the macro.
This is Asterisk 1.4.4
Doug.
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RE: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes
I previously worked for a company that did some heavy load testing with Asterisk on multiple core Sun systems. We saw that no matter how many cores you threw at Asterisk, it always used ONE core to process calls, even at very high loads. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew J. Roth Sent: Friday, June 01, 2007 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes John Hughes wrote: OpenSSI can't (at the moment) migrate threads between compute nodes. It can migrate separate processes, but doesn't Asterisk use threads? John, Asterisk uses 1 thread per call, plus about 10 to 15 background threads that persist throughout the life of the process. I'm curious if the 1 thread per call model is efficient as the number of calls increases. It's possible that in the 100+ call range that there is a significant overhead to managing all of those threads without much gain since most servers have 1 to 8 processors to actually schedule them on. Acquiring locks on shared resources between the threads could be pretty nasty at that point, too. I wonder if pooling the calls in X threads, where X is a value that is determined at compile time by looking at the number of processors available, would be more efficient? This is probably just an academic question, because I'd imagine it would require an overhaul of the codebase to accomplish. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk mysql support
Speaking of SQLite, is there an Asterisk SQLite command? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, June 01, 2007 9:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk mysql support On Fri, Jun 01, 2007 at 10:37:07AM -0500, Diego Quintana Cruz wrote: Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. The module was indeed not there. Building it. Thanks for the note, Diego. (We were focusing a bit more on sqlite recently) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] High Port Count ATA
I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] High Port Count ATA
Cory, I'm not quite clear on that. Do these channels banks have an IP uplink port so that each FXS port can SIP register to asterisk? Doug. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, May 31, 2007 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] High Port Count ATA Channel banks would work. Rhino works well, or if you need more chassis density, try the Carrier Access ADIT600 configured with FXS blades. Cory J Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, May 31, 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] High Port Count ATA I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Multiple Network Interfaces
I have a scenario here with IP phones, on a private 192.168 network connecting to an Asterisk box, also on the same 192.168 private network. We'd like to have the Asterisk box also be able to send traffic to the public IP space. For this, we would need to multi-home the box, and put two network cards in it, with two IP addresses, one on each network. I know from past experience that Asterisk only listens on the first interface, or a single one if specified. I imagine this will cause all sorts of problems with a multi homed approach. Has anyone gotten around this? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ITSP that honors Dial Around Compensation
All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: Dougs Payphone sip:[EMAIL PROTECTED];isup-oli=70;tag=as6fbc6e76 When going out through level 3, they sometimes strip the isup-oli tag. I've tried 3 other ITSP's, by calling the MCI test number 18889996365, and they must be stripping it as well, because they have all also failed. Anyone used an ITSP that passes isup-oli all the way though to TDM? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling DBQuery
Has anyone tried to compile the current version of MySQLPool from http://www.yosd.at http://www.yosd.at/ against Asterisk 1.4.4? It seems to not compile... [EMAIL PROTECTED] res_mysqlpool]# make gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -I/usr/local/mysql/include/mysql -D_REENTRANT -D_GNU_SOURCE -O6-DUSE_CVS -fPIC -c -o res_mysqlpool.o res_mysqlpool.c In file included from res_mysqlpool.c:35: res_mysqlpool.h:54: warning: function declaration isnât a prototype res_mysqlpool.c:87: warning: data definition has no type or storage class res_mysqlpool.c:87: warning: type defaults to âintâ in declaration of âSTANDARD_LOCAL_USERâ res_mysqlpool.c:89: warning: data definition has no type or storage class res_mysqlpool.c:89: warning: type defaults to âintâ in declaration of âLOCAL_USER_DECLâ res_mysqlpool.c:199: warning: function declaration isnât a prototype res_mysqlpool.c: In function âget_host_listâ: res_mysqlpool.c:648: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c:649: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c:656: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c:675: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c:680: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c:686: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c:695: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c:703: warning: assignment discards qualifiers from pointer target type res_mysqlpool.c: At top level: res_mysqlpool.c:946: warning: no previous prototype for âunload_moduleâ res_mysqlpool.c: In function âunload_moduleâ: res_mysqlpool.c:957: error: âSTANDARD_HANGUP_LOCALUSERSâ undeclared (first use in this function) res_mysqlpool.c:957: error: (Each undeclared identifier is reported only once res_mysqlpool.c:957: error: for each function it appears in.) res_mysqlpool.c: At top level: res_mysqlpool.c:964: warning: no previous prototype for âload_moduleâ res_mysqlpool.c:985: warning: no previous prototype for âreloadâ res_mysqlpool.c:992: warning: no previous prototype for âdescriptionâ res_mysqlpool.c:997: warning: no previous prototype for âusecountâ res_mysqlpool.c: In function âusecountâ: res_mysqlpool.c:999: warning: implicit declaration of function âSTANDARD_USECOUNTâ res_mysqlpool.c: At top level: res_mysqlpool.c:1004: warning: function declaration isnât a prototype make: *** [res_mysqlpool.o] Error 1 Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue cmd option 'i'
-Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Monday, January 15, 2007 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue cmd option 'i' On 1/15/07, James Fromm [EMAIL PROTECTED] wrote: Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? You're not misinterpreting. If it crashes, please file a bug at bugs.digium.com. Thanks. I wonder how this could actually work? If Asterisk sends an INVITE to a phone, and the phone responds with 'Moved Temporarily', and Asterisk sends the INVITE again, isn't the phone just going to send 'Moved Temporarily' again? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] G729 license counting
That's not correct. You need one G729 license for each transcoding instance. If you have two SIP channels and both are G729, then no license is required. If you have two SIP channels, and one is G729 and the other is ulaw, then a license is required. Doug. -Original Message- From: Zoa [mailto:[EMAIL PROTECTED] Sent: Monday, January 08, 2007 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 license counting Yes Zoa Michel wrote: Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Voicemail personalised greetings using DB/IMAPbackend?
Does this model give you functioning mwi? -Original Message- From: Ray Jackson [mailto:[EMAIL PROTECTED] Sent: Friday, January 05, 2007 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail personalised greetings using DB/IMAPbackend? Hi all, I am attempting to build a horizontally scalable Asterisk deployment and am getting very close to achieving that goal. With Asterisk 1.4 I now have an IMAP backend for Voicemail messages which is great as users can check the same messages either through the voice portal or using Webmail. However, I'm not sure the best way of dealing with personalised greetings such as a user's unavailable/busy message etc. Despite the IMAP backend these greetings appear to be stored on the local file system under /var/spool/asterisk/voicemail/default, which means if I build a farm of Asterisk servers - each will have it's own spool directory. My aim is to have *nothing* stored locally at all... If there a way of storing these greetings in a database table or using IMAP? I saw the ODBC voicemail storage module, but I would prefer to stick with a REALTIME/IMAP backend? If I mount the /var/spool/asterisk/voicemail directory remotely using a shared NFS mount on a NAS device will this work okay or lead to problems/race conditions etc.? Any advice would be welcome! Regards, Ray ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Richard, We have underscores all over the place in our config files, including others in queues.conf. I don't think that's the murder weapon. I think, in general, queues are one of Asterisks biggest features, and also one of it's shakiest. The reload, which is run from a script, caused a reload on 3 servers that are supposed to be redundant, and each crapped it's pants in a slightly different manner. The first stopped processing all queue calls (ie calls would lockup), the second core dumped, and the third seemed ok until you did another 'reload app_queue.so' where it would tell you that the previous reload was not finished yet. Someone made a post yesterday about doing 200 queues on Asterisk. I don't envy what he is about to endure. Doug. -Original Message- From: Richard Lyman [mailto:[EMAIL PROTECTED] Sent: Thursday, January 04, 2007 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen? Douglas Garstang wrote *snipped cat = 0x81507e0 mcao_QMain tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds *snipped a quick run through of of app_queue.c (my copy) for anything directly dealing with a reload shows tmp in use for realtime later a reference for convert to dashes from uunderscores i would do a quick test of a queue name without underscores ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Anyone seen this? It ocurred on a 'reload app_queue.so' command.
Asterisk version is 1.2.9.1.
Tried again, but it was not immediately reproducable.
Doug.
(gdb) bt
#0 reload_queues () at app_queue.c:3339
#1 0xb778a7a8 in reload () at app_queue.c:4012
#2 0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at
loader.c:257
#3 0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147
#4 0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out
of bounds) at cli.c:1364
#5 0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927
#6 0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305
#7 0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401
#8 0xb7f15ed8 in pthread_start_thread () from /lib/libpthread.so.0
#9 0xb7e147ea in clone () from /lib/libc.so.6
(gdb) bt full
#0 reload_queues () at app_queue.c:3339
q = (struct ast_call_queue *) 0x81adca8
ql = (struct ast_call_queue *) 0xbddfaec0
qn = (struct ast_call_queue *) 0xb7dc03b3
cfg = (struct ast_config *) 0x81aca30
cat = 0x81507e0 mcao_QMain
tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds
var = (struct ast_variable *) 0x811e340
prev = (struct member *) 0x8101b79
cur = (struct member *) 0x2854554f
newm = (struct member *) 0x0
new = 0
general_val = 0x2854554f Address 0x2854554f out of bounds
interface = '\0' repeats 79 times
penalty = 900
#1 0xb778a7a8 in reload () at app_queue.c:4012
No locals.
#2 0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at
loader.c:257
m = (struct module *) 0x81f3b10
reloaded = 2
oldversion = 863401873
reload = (int (*)(void)) 0xb778a7a0 reload
#3 0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147
x = 1
res = 1836020304
#4 0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out
of bounds) at cli.c:1364
argv = {0x8137cc0 reload, 0x8137cc7 app_queue.so, 0x0, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0xbddfa49c h¥ß½ïÀÛ·h}\\bh}\\b,
0xb7dc3fea
ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n,
0xb7e6fa00 ,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0xb7e6dff4
\034\020, 0x26 Address 0x26 out of bounds,
0x27 Address 0x27 out of bounds, 0xbddfa568 \200, 0xb7dbc0ef
\213U\b\213\002\205Àu\b\213\205pÿÿÿ\211\002ÆD\aÿ,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0x26 Address 0x26 out of bounds, 0x8227d68 ;\\b¬úæ·: 2007-01-03
15:17:39.165755\r\n, 0x0,
0x26 Address 0x26 out of bounds, 0xfbad8000 Address 0xfbad8000 out of
bounds,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d8e ,
0x8227dcc , 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0x8227dcc , 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·, 0xb700 Address 0xb700 out of
bounds, 0x0, 0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·,
0xb7e6dff4 \034\020, 0x0, 0xb7e6da00 , 0x0, 0xb7f1a756 \201Ã\236H,
0xb7f1eff4 tî, 0xb7e6fa00 , 0xb7e6fa00 ,
0xbddfa54c h¥ß½ê?Ü·, 0xb7f170eb
ëÃ\213\203pÿÿÿ;(r\022\213\203Ðÿÿÿ;(s\b\213\203¤ÿÿÿë½\213\203 ÿÿÿ\213,
0xb7e6fa10 ,
0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xb7e6fa00
, 0xb7e6dff4 \034\020, 0xb7e6dff4 \034\020,
0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xbddfa568
\200,
0xb7dc3fea
ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n,
0xb7e6fa00 ,
0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0x21
Address 0x21 out of bounds,
0x21 Address 0x21 out of bounds, 0x81ead18 \017, 0x80 Address 0x80 out
of bounds,
0x8091ffb \213\\$\030\203Ä\034ÃÇ\004$\004}
e = (struct ast_cli_entry *) 0x81197a0
x = 2
dup = 0x8137cc0 reload
tws = 0
#5 0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927
No locals.
#6 0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305
ret = 0
eqe = (struct eventqent *) 0x0
action = Command, '\0' repeats 72 times
tmp = (struct manager_action *) 0x8144818
idText = ActionID: 2007-01-03 15:17:39.165755\r\n, '\0' repeats 217
times
iabuf = 216.187.141.250
#7 0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401
m = {hdrcount = 3, headers = {Action: Command\000\n, '\0' repeats
238 times,
Command: reload app_queue.so\000\n, '\0' repeats 225 times,
ActionID: 2007-01-03 15:17:39.165755\000\n, '\0' repeats 217 times,
\000\n, '\0' repeats 253 times,
'\0' repeats 255 times repeats 76 times}}
RE: [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Bugger. :(
-Original Message-
From: Douglas Garstang
Sent: Wed 1/3/2007 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject:[asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Anyone seen this? It ocurred on a 'reload app_queue.so' command.
Asterisk version is 1.2.9.1.
Tried again, but it was not immediately reproducable.
Doug.
(gdb) bt
#0 reload_queues () at app_queue.c:3339
#1 0xb778a7a8 in reload () at app_queue.c:4012
#2 0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at
loader.c:257
#3 0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147
#4 0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out
of bounds) at cli.c:1364
#5 0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927
#6 0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305
#7 0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401
#8 0xb7f15ed8 in pthread_start_thread () from /lib/libpthread.so.0
#9 0xb7e147ea in clone () from /lib/libc.so.6
(gdb) bt full
#0 reload_queues () at app_queue.c:3339
q = (struct ast_call_queue *) 0x81adca8
ql = (struct ast_call_queue *) 0xbddfaec0
qn = (struct ast_call_queue *) 0xb7dc03b3
cfg = (struct ast_config *) 0x81aca30
cat = 0x81507e0 mcao_QMain
tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds
var = (struct ast_variable *) 0x811e340
prev = (struct member *) 0x8101b79
cur = (struct member *) 0x2854554f
newm = (struct member *) 0x0
new = 0
general_val = 0x2854554f Address 0x2854554f out of bounds
interface = '\0' repeats 79 times
penalty = 900
#1 0xb778a7a8 in reload () at app_queue.c:4012
No locals.
#2 0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at
loader.c:257
m = (struct module *) 0x81f3b10
reloaded = 2
oldversion = 863401873
reload = (int (*)(void)) 0xb778a7a0 reload
#3 0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147
x = 1
res = 1836020304
#4 0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out
of bounds) at cli.c:1364
argv = {0x8137cc0 reload, 0x8137cc7 app_queue.so, 0x0, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0xbddfa49c h¥ß½ïÀÛ·h}\\bh}\\b,
0xb7dc3fea
ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n,
0xb7e6fa00 ,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0xb7e6dff4
\034\020, 0x26 Address 0x26 out of bounds,
0x27 Address 0x27 out of bounds, 0xbddfa568 \200, 0xb7dbc0ef
\213U\b\213\002\205Àu\b\213\205pÿÿÿ\211\002ÆD\aÿ,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0x26 Address 0x26 out of bounds, 0x8227d68 ;\\b¬úæ·: 2007-01-03
15:17:39.165755\r\n, 0x0,
0x26 Address 0x26 out of bounds, 0xfbad8000 Address 0xfbad8000 out of
bounds,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68
;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d8e ,
0x8227dcc , 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n,
0x8227dcc , 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·, 0xb700 Address 0xb700 out of
bounds, 0x0, 0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·,
0xb7e6dff4 \034\020, 0x0, 0xb7e6da00 , 0x0, 0xb7f1a756 \201Ã\236H,
0xb7f1eff4 tî, 0xb7e6fa00 , 0xb7e6fa00 ,
0xbddfa54c h¥ß½ê?Ü·, 0xb7f170eb
ëÃ\213\203pÿÿÿ;(r\022\213\203Ðÿÿÿ;(s\b\213\203¤ÿÿÿë½\213\203 ÿÿÿ\213,
0xb7e6fa10 ,
0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xb7e6fa00
, 0xb7e6dff4 \034\020, 0xb7e6dff4 \034\020,
0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xbddfa568
\200,
0xb7dc3fea
ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n,
0xb7e6fa00 ,
0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0x21
Address 0x21 out of bounds,
0x21 Address 0x21 out of bounds, 0x81ead18 \017, 0x80 Address 0x80 out
of bounds,
0x8091ffb \213\\$\030\203Ä\034ÃÇ\004$\004}
e = (struct ast_cli_entry *) 0x81197a0
x = 2
dup = 0x8137cc0 reload
tws = 0
#5 0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927
No locals.
#6 0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305
ret = 0
eqe = (struct eventqent *) 0x0
action = Command, '\0' repeats 72 times
tmp = (struct manager_action *) 0x8144818
idText = ActionID: 2007-01-03 15:17:39.165755\r\n, '\0' repeats 217
times
iabuf = 216.187.141.250
#7 0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401
m = {hdrcount = 3, headers = {Action: Command\000\n
[asterisk-users] Music On Hold Between Servers
Can someone tell me how Asterisk handles music-on-hold between servers? Documentation for this is non-existent. Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2. 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both? 2. Is the situation any different if the 'trunk' between pbx1 and pbx2 is SIP or IAX? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 601 Contacts List
I don't think that's possible. We have the same issue. -Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 601 Contacts List Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the -directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in the order that I specified and NOT in alphabedical order. Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Agent presence
You could put together a web page that talks to the Asterisk Manager. -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Agent presence Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on pause. I'm using chan_agent for the agents, so agents are logged in and out using AgentCallbackLogin (I know it's deprecated in 1.4, but it's working well for us at the moment) and the agents are put on pause using PauseQueueMember and UnpauseQueueMember. I've figured out I can show whether an agent is logged in or out by creating a dummy extension with a hint as follows:- exten = 151,1,Dial(Agent/151) exten = 151,hint,Agent/151 X-Lite quite happily shows the agent as Ready when they're logged in, unavailable when logged out and On the Phone when (funnily enough) they're taking a call. However, when the agent is on pause, they are still shown as Ready. Is this a limitation of chan_agent, Pause/UnpauseQueueMember, Asterisk 1.2's presence support, or is there something else I can do in order to get the agent shown indicated as something other than Ready when they're on pause? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Agent presence
Wasn't Olle Johansen working on something that would allow (polycom phones at least) to show the status of agents on the phone... -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Agent presence Not quite the solution I was looking for - I was wanting the agent's status to be reflected in it's presence hint. I'm somewhat inclined to believe that 1.2 isn't going to do the job at this stage since I don't think it supports SIP presence to the degree required. Douglas Garstang wrote: You could put together a web page that talks to the Asterisk Manager. -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Agent presence Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on pause. I'm using chan_agent for the agents, so agents are logged in and out using AgentCallbackLogin (I know it's deprecated in 1.4, but it's working well for us at the moment) and the agents are put on pause using PauseQueueMember and UnpauseQueueMember. I've figured out I can show whether an agent is logged in or out by creating a dummy extension with a hint as follows:- exten = 151,1,Dial(Agent/151) exten = 151,hint,Agent/151 X-Lite quite happily shows the agent as Ready when they're logged in, unavailable when logged out and On the Phone when (funnily enough) they're taking a call. However, when the agent is on pause, they are still shown as Ready. Is this a limitation of chan_agent, Pause/UnpauseQueueMember, Asterisk 1.2's presence support, or is there something else I can do in order to get the agent shown indicated as something other than Ready when they're on pause? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 601 Contacts List
The directory file has an sd Speed dial index tag. The phone honours this index when displaying entries on the LCD screen and when the up arrow is pressed. However, it does not honor this order, and instead displays entries in alphabetical order, when you press the 'Directories' button. -Original Message- From: Jonathan k. Creasy [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 601 Contacts List There is an index in the configuration file which I believe it will obey. I'll try and find it later if you haven't found it by the time I get to the office. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, December 27, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 601 Contacts List I don't think that's possible. We have the same issue. -Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 601 Contacts List Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the -directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in the order that I specified and NOT in alphabedical order. Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Searching the list
You can only search a month at a time... :( -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Searching the list Mark Greene wrote: Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) http://lists.digium.com/mailman/listinfo/ Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Question about MWI in Asterisk 1.4.0
Sounds great. What's the mechanism by which Asterisk servers communicate the mwi status between them? -Original Message- From: Jean-Yves Avenard [mailto:[EMAIL PROTECTED] Sent: Monday, December 25, 2006 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about MWI in Asterisk 1.4.0 Hi On 12/26/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: No, Asterisk 1.4 does not include any functionality for multi-server MWI. The SIP functionality improvements are just better support for the 'pull' model of SIP MWI, in addition to the 'push' model Asterisk has used in the past. If I adapt the patch for multi-server WMI for Asterisk 1.4, is there any chances it would be committed to trunk? Would be ace if it became a standard feature... JY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Subscription Bug?
Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to xxx.yyy.142.139:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP xxx.yyy.142.139;branch=z9hG4bKd22096a5A22CE654;received=xxx.yyy.142.139 From: Foo Law sip:[EMAIL PROTECTED];tag=1AB6AFEA-D777BDB3 To: sip:[EMAIL PROTECTED];tag=as6ac26084 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 This is mighty strange, given this: hermes*CLI sip show peer 2529266 hermes*CLI * Name : 2529266 Secret : Set MD5Secret: Not set Context : bell_CallStart Subscr.Cont. : bell_WatchBLF Language : en Accountcode : 2529266 Asterisk is saying that bell_CallStart doesn't exist (which it doesn't), but because of that decides to not accept the SIP subscription. The two are not realated to one another. I'm wondering what Asterisk has been smoking over the last few days while I was away... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Subscription Bug?
Well there's ya problem. If 2943110 doesn't have a match in the dialplan anywhere, Asterisk pukes. What's up with that? I don't see why that is necessary. Doug. -Original Message- From: Douglas Garstang Sent: Tuesday, December 26, 2006 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Subscription Bug? Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to xxx.yyy.142.139:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP xxx.yyy.142.139;branch=z9hG4bKd22096a5A22CE654;received=xxx.yyy.142.139 From: Foo Law sip:[EMAIL PROTECTED];tag=1AB6AFEA-D777BDB3 To: sip:[EMAIL PROTECTED];tag=as6ac26084 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 This is mighty strange, given this: hermes*CLI sip show peer 2529266 hermes*CLI * Name : 2529266 Secret : Set MD5Secret: Not set Context : bell_CallStart Subscr.Cont. : bell_WatchBLF Language : en Accountcode : 2529266 Asterisk is saying that bell_CallStart doesn't exist (which it doesn't), but because of that decides to not accept the SIP subscription. The two are not realated to one another. I'm wondering what Asterisk has been smoking over the last few days while I was away... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Subscription Bug?
To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. -Original Message- From: Joshua Colp [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Subscription Bug? Douglas Garstang wrote: Well there's ya problem. If 2943110 doesn't have a match in the dialplan anywhere, Asterisk pukes. What's up with that? I don't see why that is necessary. Doug. I'm slightly confused by what you mean... can you elaborate more? -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Subscription Bug?
Asterisk, imho, should still accept the subscription request from user A. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Subscription Bug? On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. Yes, because the subscribe is against an extension, which is translated to a SIP (or other technology) user via the 'Hint' entry for that extension in the dialplan. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk crashed
Don't bother. If the version of asterisk the crash ocurred in isn't the latest,
the moderators will close the bug.
-Original Message-
From: Vicky [mailto:[EMAIL PROTECTED]
Sent: Friday, December 22, 2006 6:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk crashed
Post this at bugs.digium.com along with some more info like if it crashes at
use of some specific application or randomly .
On 22/12/06, Edwin Lam [EMAIL PROTECTED] wrote:
our * crashed twice in a month with segmentation fault
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6
#1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6
#2 0xb7e17090 in strdup () from /lib/tls/libc.so.6
#3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879
#4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at
res_musiconhold.c:180
#5 0x080673ae in ast_deactivate_generator (chan=0x9455ca0) at channel.c:1382
#6 0x08068d4e in generator_force (data=0x9455ca0) at channel.c:1405
#7 0x08061c50 in ast_read (chan=0x9455ca0) at channel.c:1857
#8 0x08069293 in ast_generic_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c)
at channel.c:3524
#10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0,
config=0xb6c4feb0) at res_features.c:1319
#11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0,
peerflags=0xb6c50568) at app_dial.c:1577
#12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at
app_dial.c:1619
#13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900,
context=0xb659fe20 op05_x, exten=0xb659ff14 00116,
priority=1, label=0x0, callerid=0x0, action=0) at pbx.c:553
#14 0x0808efea in __ast_pbx_run (c=0xb659fcd0) at pbx.c:2227
#15 0x0808fcdf in pbx_thread (data=0xb7ed1900) at pbx.c:2514
#16 0xb7f7cb63 in start_thread () from /lib/tls/libpthread.so.0
#17 0xb7e7718a in clone () from /lib/tls/libc.so.6
another one:
#0 0xb6ff38e2 in decodeMP3 () from /usr/lib/asterisk/modules/format_mp3.so
#1 0xb6ff4be6 in key () from /usr/lib/asterisk/modules/format_mp3.so
#2 0xb6ff4545 in key () from /usr/lib/asterisk/modules/format_mp3.so
#3 0x0806d3a1 in ast_readframe (s=0xb7eb490c) at file.c:570
#4 0xb7b0c134 in moh_files_generator (chan=0xb6b26dc0, data=0xb6b03328, len=0,
samples=160) at res_musiconhold.c:246
#5 0x08068cfe in generator_force (data=0xb6b26dc0) at channel.c:1401
#6 0x08061c50 in ast_read (chan=0xb6b26dc0) at channel.c:1857
#7 0x08069293 in ast_generic_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#8 0x080655fd in ast_channel_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c)
at channel.c:3524
#9 0xb78ddd29 in ast_bridge_call (chan=0xb6b26dc0, peer=0x8699fe8,
config=0xb6677eb0) at res_features.c:1319
#10 0xb7033301 in dial_exec_full (chan=0xb6b26dc0, data=0xb6677eb0,
peerflags=0xb6678568) at app_dial.c:1577
#11 0xb7031dc5 in dial_exec (chan=0x48, data=0x48) at app_dial.c:1619
#12 0x0808e445 in pbx_extension_helper (c=0xb6b26dc0, con=0x48,
context=0xb6b26f10 op05_x, exten=0xb6b27004 00116, priority=1,
label=0x0, callerid=0x0, action=0) at pbx.c:553
#13 0x0808efea in __ast_pbx_run (c=0xb6b26dc0) at pbx.c:2227
#14 0x0808fcdf in pbx_thread (data=0x48) at pbx.c:2514
#15 0xb7f5fb63 in start_thread () from /lib/tls/libpthread.so.0
#16 0xb7e5a18a in clone () from /lib/tls/libc.so.6
here's the versions of various components:
asterisk: 1.2.7.1, zaptel: 1.2.5, libpri: 1.2.2
any clues would be appreciated?
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
search=0xD6506D20
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RE: [asterisk-users] more than 32 callgroups pickupgroups
I'm no C programmer, but is this 32 limit just an array definition somewhere? Wouldn't it be a no brainer to track it down and increase it so some very large number? -Original Message- From: John Harragin [mailto:[EMAIL PROTECTED] Sent: Thursday, December 21, 2006 11:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] more than 32 callgroups pickupgroups callgroups pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
-Original Message- From: Richard Lyman [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, dialplan Douglas Garstang wrote: -Original Message- From: David Gomillion [mailto:[EMAIL PROTECTED] *snipped David, this is completely different from what I am trying to do. Let's try this a different way. Let's say you have two companies. When someone calls a number in their own company, we use their INTERNAL caller id. When they call someone in another company, we want to send their EXTERNAL caller id. How would you do this? Doug. if it is just callerid then wouldn't the gf stuff (if it still exists) work? it was something like (man i'm getting old, looking up in wiki) exten = s,1,Answer() exten = s,,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx) exten = s,2,Set(CALLERID(name)=INSIDE NAME|CALLERID(num)=xx) exten = s,3,Dial(yadda) would obviously be the callerid num of the internal exten I don't think that scales to hundreds of companies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 10:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On Tue, Dec 19, 2006 at 05:19:57PM -0700, Douglas Garstang wrote:
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
Please correct me if I'm misunderstanding your
requirements, but see
below (inline) for what I would do:
-Original Message-
[snip]
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = coo1_OffNet
[coo1_OffNet]
exten = _X.,1,Set(CALLERID(NUM)=3254000)
exten = _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten = _X.,3,Goto(syst_OffNet,${EXTEN},1)
Bradley, If I do this, then I can no longer continue with further
extensions in my dialplan as Asterisk has already matched a number.
An explicit WaitExten?
No I don't want the user to have to enter another number. Processing should
continue with the original number dialled.
*sigh*
Doug.
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 6:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG So, in the event that the logic flows beyond
DG coo1_OnNet, we want to reset the caller id of say, 3254001 Doug,
DG to 3254000 Widgets Inc.
DG exten = 3254101,1,Dial(SIP/3254101,20,tr)
DG exten = 3254102,1,Dial(SIP/3254102,20,tr)
DG exten = 3254103,1,Dial(SIP/3254103,20,tr)
[coo1_CallStart]
include = coo1_OnNet
You want something which executes here, if coo1_OnNet didn't match?
exten = _.,1,Set(CALLERID(all)=Widgets Inc 3254001)
will do that.
If you then want to continue in priority 1 instead of 2, you just do
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2]
include = syst_OnNet
include = syst_OffNet
That won't do it. Processing will continue in the current extension priority. I
need it to continue looking for an extension to match against. Once Asterisk
has matched the dialled number against an extension in the dialplan, your stuck
in an extension you can never get out and get Asterisk to go back to
looking for extensions to match against.
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RE: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
Yes, we have issues with this application being removed as well. In my opinion, it's a loss of functionality. -Original Message- From: Markus Bönke [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 6:40 AM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [asterisk-users] AgentCallbackLogin() deprecated in 1.4 Hello all, I've seen that the application AgentCallbackLogin()has been set to deprecated in version 1.4. So I've done some tests based on the tutorial queues-with-callback-members.txt coming with version 1.4. What's not clear for me is what is happening to agents.conf, it seems that it's no longer needed, and I have to define my agents using variables in extensions.ael. The other thing is, that show agents doesn't show me which agents are logged in and if I use show queue I can see local channels attached to a queue but no agents. For my point of view there is some functionality lost with the new concept. If I want to program a realtime display to show agentstates in queues based on the output from show queue, what's the concept to map agents to the local channels? How can I configure agents in future? Any comments regarding that topic are appreciated. Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Doug Crompton [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 8:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: Match a Numer - then continue with
dialplan
I haven't really been following this thread but doesn't the following
snipet kinda do this
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,playback(international-call)
exten = s,n,playback(please-enter-the)
exten = s,n,read(number,number)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,n,Macro(failann,${DIALSTATUS})
This matches 011 then could do any number of things. Here I just goto,
then it looks for more numbers (the announcement is optional) and then
dials them.
Maybe not what you are looking for but it is an example of Asterisk
matching an extension and then going on to take more digits that then
branch based on other digits. Here the 011 is prepended to the final
number.
Don't get offended Doug, but I get really frustrated when I try to explain what
I am trying to do with Asterisk, and people don't seem to quite get it. Your
about the 4th person who's replied to this post, and hasn't quite grasped my
question. :) --- smiley.. see...we're all cool.
I don't want Asterisk to go on to ask for more digits. I want to do a very
simple thing. I want to set a variable when call flow continues beyond a
certain point (without asking the user for more digits), and then continue on,
and use that variable later. It's a very simple thing, I can't work out why
Asterisk doesn't let me do that.
Surely other people have hit the situation where they first check extensions
within a company, and then if there's no match, you glue all the other
companies dialplans together with this one. At that point, when one company
dials another, the caller id that's sent should be the company caller id, not
the caller id of the individual extension. It's a very common business
requirement... at least that's what my boss, who has spend many years
installing TDM pbx's tells me.
BTW - what is a numer?
A numer is a spelling mistake. I was going to change the title, but it would
have broken the thread.
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RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message-
From: Andreas Sikkema [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
[snip]
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = coo1_OffNet
[coo1_OffNet]
exten = _X.,1,Set(CALLERID(NUM)=3254000)
exten = _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten = _X.,3,Goto(syst_OffNet,${EXTEN},1)
Bradley, If I do this, then I can no longer continue with
further extensions in my dialplan as Asterisk has already
matched a number. I still need to check black/white lists,
set pic codes and rate centers, 4 digit extensions etc within
the company context. I just need to set the caller id and
then move on. If I goto over to ${EXTEN} within syst_OffNet,
I'd have to put ALL this logic within that extension, which
would mean potentiall several hundred priorities. Asterisk
really does need a way to match a number, execute some code,
and then go back to looking for extensions.
Why not do something like this (in pseudo dialplan):
matching and initial dialplan stuff
decide the outgoing callerid should change
Ok...
SetVar(outgoing_callerid=1234567)
Bzzt. In order to call SetVar, I have to match the extension dialled. When that
happens, there is NO WAY to continue searching the dialplan after that point
for another extension to match.
continue with dialplan and do all kinds of weird things
Can only continue within the current proirity... which means that at this
point, all my further logic has to be coded as priorities in the extension that
called SetVar. Seeing as though I have several dozen more contexts to include,
this isn't feesible.
Set(CALLERID(NUM)=${outgoing_callerid})
Dial(outgoing destination)
This will not screw up your extesnions matching, but you will
need to check that outgoing_callerid has been filled before setting
callerid (or make sure it is always filled with something sensible).
Thanks for trying.
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RE: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
-Original Message- From: Gavin Hamill [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 7:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4 On Wed, 20 Dec 2006 14:39:42 +0100 Markus Bönke [EMAIL PROTECTED] wrote: Hello all, The other thing is, that show agents doesn't show me which agents are logged in and if I use show queue I can see local channels attached to a queue but no agents. For my point of view there is some functionality lost with the new concept. Funny. I said the same thing in this list about 2 months ago and I got told I was nuts. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message- From: Andreas Sikkema [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Match a Numer - then continue with dialplan Bzzt. In order to call SetVar, I have to match the extension dialled. When that happens, there is NO WAY to continue searching the dialplan after that point for another extension to match. You can't use a generic extension and search a database table for $EXTEN - callerid relation and then set it? Yes, I can do that. However, in order to do all that, I have to match an extension first. Same problem as before. Your diallingplan is _so_ different to what we do, yet what you want to do is pretty much the same to what we do all the time. I dunno about that. I think we're the only crazy ones offering company masked caller id, or else there'd be lots of people asking how to do it. But our Asterisk boxes have _no_ sip CPE's registered to them and our diallingplan is littered with database lookups. We have no static stuff in our dialingplan. And we have quite a number of users. If you have no statuc stuff in your dialplan, how do you use the 'include =' statement? We don't have users... we have companies. It's a hosted IPT service... and to make the problem even more insane, each company has multiple levels of organisational structure. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Match a Numer - then continue with
dialplan
On 20/12/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Bzzt. In order to call SetVar, I have to match the
extension dialled. When that happens, there is NO WAY to
continue searching the dialplan after that point for another
extension to match.
Can you not use either Goto or the Local channel, maybe a combination,
to restart the dialplan with your variable set? (Might need a _ or two
on the variable name to get it to survive)
The Goto() command requires priority (extension, context). I'd need to jump to
a context, without supplying an extension, which it won't accept. If I pass a
priority, we're right back at square one, we're I'm stuck in a priority and
can't get back to an extension. I tried putting a Dial(Local/${EXTEN}), but the
problem was that Asterisk then went into an infinite when I tried to include
all the company contexts together (because it was matching the Dial/Local
again).
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RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan Douglas Garstang wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Match dialed digits of 668 exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,More Stuff Here/ Ugh. 'More Stuff Here' isn't what I need Eric. I need to continue the dialplan. I need do be able to continue to search for extensions. All I want to do is set the callerid, so that later on, when we find a match, the extension can be dialled with the new caller id already set. This ain't gonna work... exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message- From: Douglas Garstang Sent: Wednesday, December 20, 2006 10:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Match a Numer - then continue with dialplan -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan Douglas Garstang wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Match dialed digits of 668 exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,More Stuff Here/ Ugh. 'More Stuff Here' isn't what I need Eric. I need to continue the dialplan. I need do be able to continue to search for extensions. All I want to do is set the callerid, so that later on, when we find a match, the extension can be dialled with the new caller id already set. This ain't gonna work... exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist exten = 669,3,include = blacklist Dang it. My fat fingers posted too soon by mistake. As I was trying to say, This obviously won't work... exten = 669,1,Set(FNORD=bob) exten = 669,2,AGI(eris.pm) exten = 669,3,include = blacklist exten = 669,4,include = whitelist exten = 669,5,include = PIC_Code_Insertion exten = 669,6,include = Rate_Center_Insertion exten = 669,7,include = Findme/Followme ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan Douglas Garstang wrote: -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 19, 2006 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug. Have you tried using the SetVar cmd? I haven't tried it but it sounds like it might work for this. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, David David, If I call setvar, my variable will be set, but dialplan processing will stop... Then something else is wrong. SetVar will not stop dialplan processing. In 1.4, I believe SetVar() was removed. Check upgrade.txt. Use Set in 1.4 instead. I was not clear. EXTENSION processing will stop. Once you've matched an extension, and your logic is running through priorities in an extension, you no longer have the ability to search for another extension to match against. That's what I need to do. Again, when control flows beyond a certain point, ie when all calls are now known to be extra-company, we need to set the callerid to the external company id... so that later on when we dial, the caller id presented to person in the other company is correct. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 10:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
I think you're making it far too difficult.
What I do is something like this:
[outgoing]
include = internal
include = longdistance
;Always include internal first, as matches from the first include
;will be used first. This allows you to make sure your internal
;extensions don't go out your trunks.
[longdistance]
ignorepat = 9;
include = default; already included from local, but putting here for
clarity
include = local;
exten = _91XXX,1,Macro(trunkout,${EXTEN}) ;Medium Distance
exten = _91XX,1,Macro(trunkout,${EXTEN}) ;Long Distance
Then, I have:
[macro-trunkout]
exten = s,1,Set(cname=${DB(showname/${CALLERIDNUM})});
exten = s,n,Set(cnum=${DB(shownum/${CALLERIDNUM})});
exten = s,n,GotoIf($[foo${cnum} = foo]?6); //if
calling from ZAP
channel that set caller ID already
exten = s,n,Set(CALLERID(name)=${cname}|a);
exten = s,n,Set(CALLERID(number)=${cnum}|a);
exten = s,n,Dial(${TRUNK}/${ARG1:${TRUNKMSD}});
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CONGESTION,1,Congestion(30)
exten = s-CONGESTION,2,Hangup
exten = s-CANCEL,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-BUSY,2,Hangup
Why is this important? It's not. But it is fundamentally
different from
what you're asking. You want to match a partial extension dialed and
then continue appending digits. What you really need to do is
wait for
the whole number, then decide what kind of number it is, do the
processing, and send it on its way. It's just a slight change
in the way
you're thinking, because you understand that there's a class
of numbers
to treat differently. And that's OK. Just don't do anything with it
until the whole extension has been entered!
Uhm, No. I'm not trying to partially match extensions and then continue
appending digits. What makes you think that?
You'll notice that, anything not going through the trunkout macro
doesn't get tweaked, and anything that goes through there
will read from
the database. I could just as easily set a single value, but
I have some
users that I want to go out as themselves, and different departments
that have a general number, etc. I found the Asterisk
Database to be the
easiest to tweak, as I have some scripts to allow admins to
change the
effective CallerID on the fly.
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies. When someone
calls a number in their own company, we use their INTERNAL caller id. When they
call someone in another company, we want to send their EXTERNAL caller id. How
would you do this?
Doug.
I hope this helps! Asterisk can do what you're asking, and it
does every
day.
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message- From: Tony Mountifield [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 11:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Match a Numer - then continue with dialplan In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Don't get offended Doug, but I get really frustrated when I try to explain what I am trying to do with Asterisk, and people don't seem to quite get it. Your about the 4th person who's replied to this post, and hasn't quite grasped my question. :) --- smiley.. see...we're all cool. Perhaps its the terminology you used that is confusing people. See below: I don't want Asterisk to go on to ask for more digits. I want to do a very simple thing. I want to set a variable when call flow continues beyond a certain point (without asking the user for more digits), and then continue on, and use that variable later. It's a very simple thing, I can't work out why Asterisk doesn't let me do that. To almost all people call flow would mean executing one priority after another for a given extension. After reading and re-reading your posts trying to work out what you are trying to do, it seems to me that when *you* say call flow, you mean the act of trying to find an extension. And what your looking for is a way to do things a different points in the *search*, while it is still trying to decide on a statement to land on. Is that correct? Yes to the first sentence. Not quite sure what you mean after that. If so, I think you need to re-think the strategy a bit. The only way a command gets executed in a dialplan is when Asterisk has matched an extension and a priority. Then once it has executed that command, it increments the priority (unless it was a Goto or something) and starts searching again. That was my original question. I was asking if there was a way to set a variable and the continue, which doesn't seem like too strange a thing to have Asterisk support. However, don't forget that it searches for matching extensions every time the priority changes. You are not locked into a particular pattern or extension number from priority 1 onwards. You can mix and match patterns with literal extensions, even across includes, e.g. Don't follow you. When asterisk matches an extension, it starts interating through the priorities until there's none left, or you Goto() somewhere else. [example] include = ctx31X include = ctx3XX exten = _X.,1,NoOp(this gets executed first for everything) exten = _X.,2,NoOp(this gets executed second only if ctx31X or ctx3XX didnt match) exten = _X.,3,NoOp(this gets executed third for everything) You lost me here. [ctx31X] exten = _31X,2,NoOp(this gets executed second for 310-319) [ctx3XX] exten = _3XX,2,NoOp(this gets executed second for 300-309 and 320-399) So you might be able to do something along these lines by being creative with priorities and includes, and setting or testing variables. Something along these lines: 1. Each company starts off in its own context, and at Can't do that. The point at which a phone enters the dial plan needs to start with rather a long list of include= statements, to grant/deny access to certain features. priority 1 in _X. it sets a variable like SRCCOMPANY to something specific to it. It includes all the destination contexts. 2. Each destination context starts at priority 2 and sets a variable like DESTCOMPANY to something specific to that destination. 3. At priority 3 in each source context, SRCCOMPANY and DESTCOMPANY are compared, in order to decide whether to override the CallerID with the source company's generic callerID. Let's say this uses priorities 3, 4 and 5 (for the GotoIf doing the compare, then the SetCallerID, and the NoOp target for the GotoIf when the callerID doesn't need rewriting). The destination contexts do not have priorities 3, 4 and 5. 4. The destination contexts continue at priority 6 to route the call. I think by interleaving priorities between contxts like this you should be able to achieve what you are looking for. Please let us know on the list if you are successful - it encourages us to keep helping in the future! I tried your example, which I completely don't follow, and it didn't seem to execute as you expected. Dialling 311 yields: *CLI -- Executing NoOp(SIP/3254101-d10e, this gets executed first for everything) in new stack -- Executing NoOp(SIP/3254101-d10e, this gets executed second only if ctx31X or ctx3XX didnt match) in new stack -- Executing NoOp(SIP/3254101-d10e, this gets executed third for everything) in new stack I need to make extensive use of the include= directive, and I just can't see how getting stuck in priorities within an extension is going to allow me to do that. Doug
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message- From: Tony Mountifield [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 11:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Match a Numer - then continue with dialplan In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Don't get offended Doug, but I get really frustrated when I try to explain what I am trying to do with Asterisk, and people don't seem to quite get it. Your about the 4th person who's replied to this post, and hasn't quite grasped my question. :) --- smiley.. see...we're all cool. Perhaps its the terminology you used that is confusing people. See below: I don't want Asterisk to go on to ask for more digits. I want to do a very simple thing. I want to set a variable when call flow continues beyond a certain point (without asking the user for more digits), and then continue on, and use that variable later. It's a very simple thing, I can't work out why Asterisk doesn't let me do that. To almost all people call flow would mean executing one priority after another for a given extension. After reading and re-reading your posts trying to work out what you are trying to do, it seems to me that when *you* say call flow, you mean the act of trying to find an extension. And what your looking for is a way to do things a different points in the *search*, while it is still trying to decide on a statement to land on. Is that correct? If so, I think you need to re-think the strategy a bit. The only way a command gets executed in a dialplan is when Asterisk has matched an extension and a priority. Then once it has executed that command, it increments the priority (unless it was a Goto or something) and starts searching again. However, don't forget that it searches for matching extensions every time the priority changes. You are not locked into a particular pattern or extension number from priority 1 onwards. You can mix and match patterns with literal extensions, even across includes, e.g. [example] include = ctx31X include = ctx3XX exten = _X.,1,NoOp(this gets executed first for everything) exten = _X.,2,NoOp(this gets executed second only if ctx31X or ctx3XX didnt match) exten = _X.,3,NoOp(this gets executed third for everything) [ctx31X] exten = _31X,2,NoOp(this gets executed second for 310-319) [ctx3XX] exten = _3XX,2,NoOp(this gets executed second for 300-309 and 320-399) So you might be able to do something along these lines by being creative with priorities and includes, and setting or testing variables. Something along these lines: 1. Each company starts off in its own context, and at priority 1 in _X. it sets a variable like SRCCOMPANY to something specific to it. It includes all the destination contexts. I think that's the deal breaker right there. I can't start a company within an extension. The starting point for each phone within a company needs to make extensive use of the include= directive. Features will be disabled by default, so there will be a list of includes to block unpurchased features. Then we'll include contexts for 911, voicemail retrieval and general numbers, ie: [coo1_CallStart] include = syst_FeaturePersonalMeetmeBlock include = syst_FeatureIntercomBlock include = syst_FeatureIDDBlock include = syst_Emergency include = syst_VMRetrieve include = coo1_General include = syst_GeneralInternal include = syst_ExportedApps include = syst_Route Finally, when we're finished scanning for blocked services, and asterisk terminated extensions, we try to route the call from this phone to the destination number, either OnNet or OffNet. That's where syst_Route comes in. For managability, we have to use lots of includes. We can't have our entire dialplan as one big _X. extension match. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dial own extension to get to voicemail.
What about comparing the caller id to the dialled number, and if they match, then call Voicemail() ? -Original Message- From: Phil Finkler [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dial own extension to get to voicemail. I've gotten this Polycom 501 pretty much licked, but I need to know if there's a way in a dialplan to say if someone dials their own extension it goes straight to voicemail and asks them for their password. I thought I saw an example of this on the web but I can't seem to find it. Any advice appreciated! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 1:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Match a Numer - then continue with dialplan DG == Douglas Garstang [EMAIL PROTECTED] writes: DG If I pass a priority, we're right back at square one, we're I'm DG stuck in a priority and can't get back to an extension. You ALWAYS have both a priority and an extension. There is no such thing as being stuck in a priority. Benny, lets say I have this... exten = _X.,1,NoOp(1) exten = _X.,2,NoOp(2) exten = _X.,3,NoOp(3) - Current code execution location exten = 555,1,NoOp(1) exten = 555,2,NoOp(2) exten = 555,3,NoOp(3) How would I jump back into the dialplan from the current execution location and continue to search for matches? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
[example] include = ctx31X include = ctx3XX exten = _X.,1,NoOp(this gets executed first for everything) exten = _X.,2,NoOp(this gets executed second only if ctx31X or ctx3XX didnt match) exten = _X.,3,NoOp(this gets executed third for everything) [ctx31X] exten = _31X,2,NoOp(this gets executed second for 310-319) [ctx3XX] exten = _3XX,2,NoOp(this gets executed second for 300-309 and 320-399) Does this really work? I've never seen this behavior documented anywhere. Asterisk always searches the current context before looking in included ones for a start. Second, I don't see how it can just jump out of [example] into [ctx31X] and back again without being told to do so ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG Surely other people have hit the situation where they first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just Goto(gluedtogethercontext,${EXTEN},1)
After doing which, you can no longer use the include = directive.
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Benny Amorsen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG == Douglas Garstang [EMAIL PROTECTED] writes:
-Original Message- From: Benny Amorsen
[mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006
6:16 AM To: asterisk-users@lists.digium.com Subject:
[asterisk-users] Re: Match a Numer - then continue with dialplan
exten = _.,n,Goto(coo1_CallStart2,${EXTEN},1)
[coo1_CallStart2] include = syst_OnNet include = syst_OffNet
DG That won't do it. Processing will continue in the current
DG extension priority. I need it to continue looking for an extension
DG to match against. Once Asterisk has matched the dialled number
DG against an extension in the dialplan, your stuck in an
DG extension you can never get out and get Asterisk to go back to
DG looking for extensions to match against.
It looks for extensions to match against all the time. What you say
makes no sense.
E.g. this code works, with EXTEN being 321 and starting in incoming.
[incoming]
exten = _3XX,1,NoOp(We get to this place)
exten = _X2X,2,Goto(incoming,${EXTEN},700)
exten = _XX1,700,NoOp(We end up here)
If EXTEN was 301, only priority 1 would run. If it was 320, priority 1
and 2 would run.
Ok, but how does that help me? All I want to do is set a variable to be used
later on in the dialplan.
Eg, if someone dialls 2944000, which is in a different company...:
[co1_phone-start]
include = co1_did
include = sys_glue
[co1_did]
exten = 3254101,1,Dial(SIP/3254101,18,tr)
exten = 3254102,1,Dial(SIP/3254102,18,tr)
exten = 3254103,1,Dial(SIP/3254103,18,tr)
; No match, so now we want to use the external caller id variable for use later
on, when
; we finally dial the dest number after performing all restriction and feature
checks.
; Actually I just realised we want to SET the caller id.
[sys-glue]
include co1_did
include co2_did
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
DG Surely other people have hit the situation where they
first check
DG extensions within a company, and then if there's no match, you
DG glue all the other companies dialplans together with this one.
Of course we have. Just Goto(gluedtogethercontext,${EXTEN},1)
After doing which, you can no longer use the include = directive.
Perhaps I can get a clarification before proceeding further...
In reading the thread the situation seems to be: Company A
users has a
user with extension/callerid XXX, he calls someone in company
B and you
want to set the callerid to company A's main number rather than the
userr's default callerid?
Is this correct?
Mike,
Exactamundo.
Doug.
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RE: [asterisk-users] Re: Match a Numer - then continue with dialplan
-Original Message-
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with
dialplan
Perhaps I can get a clarification before proceeding further...
In reading the thread the situation seems to be: Company A
users has a
user with extension/callerid XXX, he calls someone in company
B and you
want to set the callerid to company A's main number
rather than the
userr's default callerid?
Is this correct?
Mike,
Exactamundo.
Doug.
Ok.
How about:
;outgoing context for company A
[companyA]
;Various include statements
include = foo
.
.
.
;Handle calls from A - B
;Here will match company B numbers
exten = , 1, Set(CALLERID=CompanyAMain)
exten = , 1, Dial(${EXTEN}
You can do the inverse for companyB, or you could l have a
single macro
that deals with calls to/from each company and decides what
do to based
on the callerid making the call.
Mike.
Mike, this is a hosted IPT solution. There's potentially going to be hundreds
(we hope) of companies hosted and configured on this box. I'd have to write
static code to compare every number in every company to every number in every
other company, and that's just not feesible.
Doug.
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RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
-Original Message- From: Tony Mountifield [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 2:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Match a Numer - then continue with, dialplan In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Let's try this a different way. Let's say you have two companies. When someone calls a number in their own company, we use their INTERNAL caller id. When they call someone in another company, we want to send their EXTERNAL caller id. How would you do this? Firstly, in the setup you are envisaging, how do you distinguish which company the caller is calling from? Their extensions number? The context at which they enter the dialplan? Or something else? Good questions, all of them. Unfortnately, I don't have answers to them. I wanted to take our 3000 line python script, which we'd used due to inadequacies of the dialplan, and throw the horrible nasty thing out the window. Secondly, how do you distinguish between destination numbers in one company from those in another? Number range? Context? My brain hurts. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
-Original Message- From: Richard Lyman [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 20, 2006 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, dialplan Douglas Garstang wrote: -Original Message- From: David Gomillion [mailto:[EMAIL PROTECTED] *snipped David, this is completely different from what I am trying to do. Let's try this a different way. Let's say you have two companies. When someone calls a number in their own company, we use their INTERNAL caller id. When they call someone in another company, we want to send their EXTERNAL caller id. How would you do this? Doug. if it is just callerid then wouldn't the gf stuff (if it still exists) work? it was something like (man i'm getting old, looking up in wiki) exten = s,1,Answer() exten = s,,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx) exten = s,2,Set(CALLERID(name)=INSIDE NAME|CALLERID(num)=xx) exten = s,3,Dial(yadda) would obviously be the callerid num of the internal exten If there's hundreds of companies on this box, we'd need an exponentially larger number of statements... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan
I seriously doubt he'd know how to get on the 'Internets' -Original Message- From: Doug Crompton [mailto:[EMAIL PROTECTED] Sent: Wed 12/20/2006 8:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan On Wed, 20 Dec 2006, Michael Collins wrote: After listing all of that, then give us the description of what needs to happen next, the part about deciding which caller ID info to send. Pretend like you're explaining it to a bunch of idiots who understand only small words and short sentences. :) Damn, I didn't know Bush was subscribed to this list! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is MOH Still Broken in Asterisk 1.4 (beta3)?
I'm wondering if moh is still broken in Asterisk 1.4 beta3. In Asterisk 1.2, when a callee put a caller on hold, the musiconhold class that was played was not the one the callee wanted the caller to hear, but something else. Even after using mohsuggest in Asterisk 1.4, it still appears that this is not working correctly. Here's the results of a simple test: CASE CALLER CALLEE HOLDER HOLDER HEARS MOH -- 1325410132541023254101moh1 2325410132541023254102default 3325410232541013254102moh2 4325410232541013254101default For each extension, I have mohsuggest set. Test cases 1 and 3, where the caller puts the callee on hold, yield the expected behaviour. However, test cases 2 and 4 where the callee puts the caller on hold, do not yield the correct results. Here's what the results SHOULD be. CASE CALLER CALLEE HOLDER HOLDER HEARS MOH -- 1325410132541023254101moh1 2325410132541023254102moh2 3325410232541013254102moh2 4325410232541013254101moh1 Am I possibly doing something wrong with mohsuggest? sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [3254101] type = friend context = CallStart username = 3254101 accountcode = 3254101 qualify = yes canreinvite = no host = dynamic dtmfmode = rfc2833 nat = no callerid = Douglas Garstang 3254101 secret = password mohsuggest = moh1 [3254102] type = friend context = CallStart username = 3254102 accountcode = 3254102 qualify = yes canreinvite = no host = dynamic dtmfmode = rfc2833 nat = no callerid = Douglas Garstang 3254101 secret = password mohsuggest = moh2 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Match a Numer - then continue with dialplan
Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
I just know someone is going to ask 'why would you ever want to do that?'.
Here's my answer.
We have two companies, each with a dialplan similar to what's below. In the
event that the number being dialled does not match any number within our OWN
company, we want to set the caller id to be a generic one for the company, NOT
one for the user. This is a pretty normal requirement that most companies want.
So, in the event that the logic flows beyond coo1_OnNet, we want to reset the
caller id of say, 3254001 Doug, to 3254000 Widgets Inc. If there was a way
to match against a number in the dialplan, and then continue execution after
that point, we could put this statement at the end of the coo1_OnNet context
and it would all be sweet. Without that, I don't have a clue how to do this...
unless we stick with out current 3,000 line python script.
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
[coo1_OnNet]
exten = 3254101,1,Dial(SIP/3254101,20,tr)
exten = 3254102,1,Dial(SIP/3254102,20,tr)
exten = 3254103,1,Dial(SIP/3254103,20,tr)
exten = 1000,1,Answer
exten = 1000,2,Wait(1)
exten = 1000,3,NoOp(${FOO})
[syst_OnNet]
include = coo1_OnNet
include = coo2_OnNet
[syst_OffNet]
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],180,tr)
~
-Original Message-
From: Douglas Garstang
Sent: Tuesday, December 19, 2006 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Match a Numer - then continue with dialplan
Anyone know if there's a way to match a dialplan extension,
execute some code, say set a variable, and then continue with
the dialplan?
I want to set a variable when the dialplan flows beyond a
certain context. This would be a great feature.
Doug.
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RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 19, 2006 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Match a Numer - then continue with dialplan On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote: Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug. Have you tried using the SetVar cmd? I haven't tried it but it sounds like it might work for this. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, David David, If I call setvar, my variable will be set, but dialplan processing will stop... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Match a Numer - then continue with
dialplan
Please correct me if I'm misunderstanding your requirements, but see
below (inline) for what I would do:
-Original Message-
[snip]
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet
Instead of including your system-wide logic for offnet calling,
introduce a per-company offnet and include that instead:
[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = coo1_OffNet
[coo1_OffNet]
exten = _X.,1,Set(CALLERID(NUM)=3254000)
exten = _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
exten = _X.,3,Goto(syst_OffNet,${EXTEN},1)
Bradley, If I do this, then I can no longer continue with further extensions in
my dialplan as Asterisk has already matched a number. I still need to check
black/white lists, set pic codes and rate centers, 4 digit extensions etc
within the company context. I just need to set the caller id and then move on.
If I goto over to ${EXTEN} within syst_OffNet, I'd have to put ALL this logic
within that extension, which would mean potentiall several hundred priorities.
Asterisk really does need a way to match a number, execute some code, and then
go back to looking for extensions.
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RE: [asterisk-users] Match a Numer - then continue with dialplan
Leo, sorry I completely don't follow you. I don't see how the registry (astdb) can help me here. -Original Message- From: Leo Ann Boon [mailto:[EMAIL PROTECTED] Sent: Tue 12/19/2006 6:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] Match a Numer - then continue with dialplan Douglas Garstang wrote: I just know someone is going to ask 'why would you ever want to do that?'. Here's my answer. We have two companies, each with a dialplan similar to what's below. In the event that the number being dialled does not match any number within our OWN company, we want to set the caller id to be a generic one for the company, NOT one for the user. This is a pretty normal requirement that most companies want. So, in the event that the logic flows beyond coo1_OnNet, we want to reset the caller id of say, 3254001 Doug, to 3254000 Widgets Inc. If there was a way to match against a number in the dialplan, and then continue execution after that point, we could put this statement at the end of the coo1_OnNet context and it would all be sweet. Without that, I don't have a clue how to do this... unless we stick with out current 3,000 line python script. If you're not using realtime to store your SIP registry, you should be able to look up the number in the family SIP/Registry (case sensitive) using the DB functions. If you're using realtime, then you'll have to do an SQL query. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Best way to access MySQL data from dial plan
I'm not sure that any solution with the MySQL dialplan command is going to be
ideal. You also can't nest your queries, ie the connectid/result id seems to
only be good for one resultset at a time... try doing something like
findme/followme with that!
Doug.
-Original Message-
From: kjcsb [mailto:[EMAIL PROTECTED]
Sent: Monday, December 18, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Re: Best way to access MySQL data from dial
plan
Resending as message didn't show up the first time
I need to access MySQL from the dial plan. Currently I am
using the MYSQL
function:
exten = *78,n,MYSQL(Connect asterisklocal localhost
asteriskuser password
asterisk)
exten = *78,n,MYSQL(Query resultid ${asterisklocal} CALL\
sp_ins_into_avp(\'/DND/${CALLERID(number)}\'\,\'YES\'))
exten = *78,n,MYSQL(Clear ${resultid})
exten = *78,n,MYSQL(Disconnect ${asterisklocal})
This shows authentication details in the Asterisk CLI which
is not ideal.
What is the recommended way to access MySQL data?
Asterisk 1.2
CentOS 4.4
MySQL 5.0
Regards
Cameron
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RE: [asterisk-users] Remote Reboot of a Polycom
From the Asterisk console: sip notify polycom-check-cfg ipaddr Or you might have to switch the polycom-check-cfg and the ip. I forget the order. You also need to make sure that the phone has alwaysreboot=1 in the sip.cfg xml file. Doug. -Original Message- From: Klaverstyn, David C [mailto:[EMAIL PROTECTED] Sent: Mon 12/18/2006 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:[asterisk-users] Remote Reboot of a Polycom Does anyone know how to remotely reboot a PolyCom specifically 601 phone? winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Between Asterisk Servers
Scenario: A call is sent from one Asterisk system to another with IAX. The remote Asterisk system runs the Queue application, which then starts to play a different music on hold class then the standard 'default'. The console on this system displays: -- Executing Queue(IAX2/xxx.yyy.142.203:4569-4, demo_QMain|t|||60) in new stack -- Started music on hold, class 'demo_MainOffice', on IAX2/xxx.yyy.142.203:4569-4 -- Called SIP/2943367 -- Called SIP/2943368 -- SIP/2943367-1bb8 is ringing -- SIP/2943368-537f is ringing However, on the first Asterisk system, we see this on the console: -- Called dundiapps:[EMAIL PROTECTED]/demo_EMain -- Call accepted by xxx.yyy.142.204 (format g729) -- Format for call is g729 -- Started music on hold, class 'default', on IAX2/xxx.yyy.142.203:4569-5 The music on hold class in use is not being conveyed back to the original Asterisk system. Please don't tell me this is a limitation. That would be very very bad. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom MyStat
Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom MyStat
It still has to go through the upstream pbx/proxy. Each phone doesn't know the location, ie ip address, of the other phones. When the state changes, it should send an updated SIP subscription to Asterisk. -Original Message- From: LST [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 13, 2006 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom MyStat On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug I think that is strictly a Polycom to Polycom thing (Buddywatch). I do not believe it affects Asterisk (i.e. Busy = DND). With that being said, I don't think it works very well even with all Polycom phones. I can change my status to Busy and look at the other Polycom Phones and they still show me as Online. (Yes, I have bw set to 1.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Core Dump: create_transaction (p=0x0) at pbx_dundi.c:2787
Anyone seen this...? Is it a known issue?
I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't
against the latest code I get given crap for it. Given that most of the time
you don't know HOW to reproduce a problem on the latest code anyway, not
accepting bugs from older versions does the community no service, because
potential bugs are never accepted for submission.
(gdb) bt full
#0 0xb7da8d3c in mallopt () from /lib/libc.so.6
No symbol table info available.
#1 0xb7da7e3a in malloc () from /lib/libc.so.6
No symbol table info available.
#2 0xb7b30aa1 in create_transaction (p=0x0) at pbx_dundi.c:2787
trans = (struct dundi_transaction *) 0x0
#3 0xb7b3e616 in find_transaction (hdr=0xbe9fda40, sin=0xbe9ffa40) at
pbx_dundi.c:361
trans = (struct dundi_transaction *) 0x0
#4 0xb7b3e0ef in handle_frame (h=0xbe9fda40, sin=0xbe9ffa40,
datalen=-1209714176) at pbx_dundi.c:1944
trans = (struct dundi_transaction *) 0xbe9ffa40
#5 0xb7b3b3ff in socket_read (id=0x81a61e0, fd=18, events=1, cbdata=0x0) at
pbx_dundi.c:2006
sin = {sin_family = 2, sin_port = 43025, sin_addr = {s_addr =
3415129048}, sin_zero = \000\000\000\000\000\000\000}
res = -1209714176
buf =
t¶\000\000\000\000\211\000\000\006\000\016\f¡\222M\023\004\022KûD\020PÜ\226¶
[EMAIL
PROTECTED](Yi\233TÇ\002Â8èÃ\023\231¸_\220k\0350\227QÙT\031è1ï[oþ}ý\232\\Ã\232ô\224Ægì\026ÀÀuy\231¬å¸\017Úzr)¨åëªb\000nËé5Nºaòdü0¥¦\f®R\237}GDáÄ,\201PFèµÅýÑOû\2076ß©ñ æ¨\022\200\021\202ñI%\t|H\232,m\rh}\235¥|[EMAIL
PROTECTED],¤ûcñ\216æì\214ëS\034\232\016\226449y±\031oñ\201ZÆ_«·c...
len = 16
#6 0x080558cd in ast_io_wait (ioc=0x8134128, howlong=-1209714176) at io.c:284
res = 1
x = 0
origcnt = 1
#7 0xb7b35e6f in network_thread (ignore=0x0) at pbx_dundi.c:2106
res = -1209714100
#8 0xb7ef9ed8 in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9 0xb7df87ea in clone () from /lib/libc.so.6
No symbol table info available.
(gdb)
Doug.
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