[asterisk-users] pull a call from a queue
We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco SPA504G, transfer asterisk page()
exten = 179,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = 179,2,Page(SIP/180SIP/181SIP/182SIP/184) The asterisk11 page() application works great, but I've just learned that the person who initiated the page can transfer or conference the page if they don't hang it up before using those functions. It never would have occurred to me to try it, but a user did it accidentally today and it caused quite a stir when somebody's conversation with a caller was being broadcast from every phone. They're using the conf and xfer buttons on the phone to make this happen, so I'm not sure if asterisk can even prevent them from doing it or if I have to figure out a way to stop it from happening on the phone. The i option for Page didn't help. Anybody dealt with this before? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NAT friendly settings
On 1/8/2014 4:17 AM, Ishfaq Malik wrote: On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia Have you added directmedia=no? Nope, I'll look into that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk NAT friendly settings
I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone - NAT/FIREWALL - Internet - NAT/Firewall- Asterisk
top posting is superior anyway --- *ducking to avoid thrown objects* If I recall correctly, when doing something like that with a polycom I had to set the registration interval absurdly low, like 20 seconds or something. I think the Polycom didn't send keepalives and that was the workaround. top posting so as to not make thread even more confusing. Nick, I have nat=force_rport,comedia in sip.conf. It is my understanding that nat=yes is deprecated? Thanks, JohnM On 01/02/2014 10:51 AM, Nick Olsen wrote: Make sure you have nat=yes in your sip.conf either under globals or individual sip peer settings. Nick Olsen Network Operations (855) FLSPEED x106 *From*: John Millican j...@millican.us *Sent*: Thursday, January 02, 2014 10:50 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: [asterisk-users] Phone - NAT/FIREWALL - Internet - NAT/Firewall- Asterisk Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the phone to my Cell, my cell rings once and then stops. I get a packet retransmission error: WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical Response) Followed by: n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no reply to our critical packet I am assuming that there is a problem with NAT. I have externip set in sip.conf. Any pointers to what I am missing? Thanks, JohnM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 CDR vs VoIP Innovations CDR
When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 CDR vs VoIP Innovations CDR
A fair guess May be as simple as this: When you terminate a call you start the call before they even get it. When they originate a call, they start the call before you get it. Just a guess without really thinking about this too much. On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: When I compare my total minutes on the bill from VoIP Innovations, to the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in the count of minutes. I'm wondering why it's there. Are there different methods of counting the billable start or end point of a phone call? If it matters, I'm counting more termination minutes than they are and they're counting more origination minutes than I am. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SetCallerPres questions
Does SetCallerPres(Prohib) remove the ANI data from a SIP call or does it simply set a flag telling other devices not to display the data? In other words, could another system override that and see the caller ID anyway? The answer may affect how I handle 911 calls, so I'm very curious. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip unregister ' (where is the peer name) will unregister a peer - however, I want to force registration of a peer from the CLI. Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? A REGISTER request originates from the peer. How do you propose Asterisk ask the unregistered peers to REGISTER in a device agnostic fashion? Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen at? I don't think I've seen anything that has a register command, but lots of devices can get a check your config or reboot command via SIP NOTIFY. I'm more wondering why the peer is unregistered but we still expect to communicate with it. Other than a network problem or the device being unplugged...neither of which could be fixed from the server. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MoH with message on intervals
Looks like it would be pretty darn easy if I was using a queue. I could just use the periodic announcement and fiddle with specified sound files. Sadly, I'm calling the phones with SLATrunk. The hold music will only be heard by the caller when the user pushes the hold button on their phone. I can definitely break up the hold music into segments and playing the directory with sort=alpha. I guess it won't be that hard, I was just hoping there was a built in option that I hadn't noticed :) Or you could just do a Breakout IVR if they are in a queue ... easy to manage and update. On Mon, Jan 21, 2013 at 4:43 PM, Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com wrote: The simplest way to do it would be to use sox to remix your moh file with the message like this: Let's say you're using the standard file macroform-cold_day.wav. First you split it into two minute segments like so Sox macroform-cold_day.wav seg1.wav trim 0.0 120.0 Sox macroform-cold_day.wav seg2.wav trim 0.0 120.0 Sox macroform-cold_day.wav seg3.wav trim 0.0 120.0 Now put it back together with your message inserted like this: Sox seg1.wav yourmessage.wav seg2.wav yourmessage.wav seg3.wav yourmessage.wav macroform-cold_day.wav -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett Sent: Monday, January 21, 2013 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MoH with message on intervals I'm talking to somebody who wants to have a recorded message play periodically for people on hold. An example would be interrupting the hold music every two minutes to play a message with business hours and current specials. Seems like you could fake it by breaking the music files into two minute chunks with alphabetical file names, and using sort=alpha. It seems like there might also be possible ways to do in the dialplan with 'Set(CHANNEL(musicclass)=' or a combination of StartMusicOnHold() and StopMusicOnHold(). Can anybody point me in the right direction? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MoH with message on intervals
I'm talking to somebody who wants to have a recorded message play periodically for people on hold. An example would be interrupting the hold music every two minutes to play a message with business hours and current specials. Seems like you could fake it by breaking the music files into two minute chunks with alphabetical file names, and using sort=alpha. It seems like there might also be possible ways to do in the dialplan with 'Set(CHANNEL(musicclass)=' or a combination of StartMusicOnHold() and StopMusicOnHold(). Can anybody point me in the right direction? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'Slower but cleaner' G711 option
When you compile asterisk from source there's an option to enable an alternate G711 algorithm which is stated somewhat cryptically to be slower, but cleaner. Does anybody have the authoritative answer as to what the deal is with this? I saw a forum post from somebody who said something about it handling faxes better, and only being marginally slower. If it produces better audio, and isn't much slower why isn't it the default option? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL syntax consistency
How consistent has the syntax for extensions.ael been from version to version? extensions.conf has annoyed me in this regard. i.e.: commas to pipes, pipes back to commas, macro to gosub, etc etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Too many open files: what might cause this?
So a few people just reported that they couldn't make any calls. I logged into asterisk and at first everything on the console looked normal, then I got swamped with messages about too many open files. This is from my asterisk/messages log file: [Oct 2 16:46:00] WARNING[19429] rtp.c: Unable to allocate RTCP socket: Too many open files [Oct 2 16:46:00] WARNING[19429] udptl.c: Unable to allocate socket: Too many open files [Oct 2 16:46:00] WARNING[19429] acl.c: Cannot create socket [Oct 2 16:46:00] WARNING[19429] channel.c: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n Messages like that repeat a few dozen times, and then I get this one manager.c: Accept returned -1: Too many open files ...and that repeated tens of thousands of times. I killed asterisk and restarted it. Looks normal again. What the heck just happened? A bug? Was I attacked? Maybe I'm honestly hitting some system limit and I should bump up max file descriptors like the message says? We do have a few hundred SIP peers and maybe we'll hit 20-30 simultaneous calls at peak times but I didn't think that was particularly high load. This is Asterisk 1.4.44. I know the 1.4 branch is old, but it had been trouble free for years (until now), and I'd have to rewrite some config syntax to upgrade so I didn't see a need to do it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Too many open files: what might cause this?
I was looking at open files. lsof | wc -l tells me around 2000 or so. The total number goes up and down, but hovers around 2000 and doesn't seem to show any upward trend. I haven't rebooted the system since I killed and restarted asterisk, so the first guess would be that asterisk is what had all the files open. I wish I had checked that before I killed it. cat /proc/sys/fs/file-max says 367467 so I guess whatever happened must have been pretty extreme. So a few people just reported that they couldn't make any calls. I logged into asterisk and at first everything on the console looked normal, then I got swamped with messages about too many open files. This is from my asterisk/messages log file: [Oct 2 16:46:00] WARNING[19429] rtp.c: Unable to allocate RTCP socket: Too many open files [Oct 2 16:46:00] WARNING[19429] udptl.c: Unable to allocate socket: Too many open files [Oct 2 16:46:00] WARNING[19429] acl.c: Cannot create socket [Oct 2 16:46:00] WARNING[19429] channel.c: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n Messages like that repeat a few dozen times, and then I get this one manager.c: Accept returned -1: Too many open files ...and that repeated tens of thousands of times. I killed asterisk and restarted it. Looks normal again. What the heck just happened? A bug? Was I attacked? Maybe I'm honestly hitting some system limit and I should bump up max file descriptors like the message says? We do have a few hundred SIP peers and maybe we'll hit 20-30 simultaneous calls at peak times but I didn't think that was particularly high load. This is Asterisk 1.4.44. I know the 1.4 branch is old, but it had been trouble free for years (until now), and I'd have to rewrite some config syntax to upgrade so I didn't see a need to do it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'Training mode'
I was asked today if we could somehow have a trainee on the phone with a supervisor conferenced in, but somehow have it so anything the supervisor says is only heard by the trainee and not the customer. Is there a feature like that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk distributions
I'm looking at replacing a PBX for a small business with an asterisk box. I'm rather attracted to the idea of one of the iso distributions where someone did most of the integration for us already ;) Can anyone comment on the pros/cons of the various options? I'm seeing several options out there: -Trixbox CE (no new version since 2010? is this project dead?) -Asterisk NOW -PBX in a Flash -Elastix -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk distributions
Are there any particular reasons anybody would cite to choose one over the other? FreePBX have also an ISO distribution - I would recommend to use that one. HTH, Ioan On Wed, Feb 29, 2012 at 7:43 PM, Danny Nicholasda...@debsinc.com wrote: Asterisk Now should serve your needs nicely. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. I am probably splitting hairs, but that's not always true because there's no guarantee that the reply traveled the same path as the echo request. If you dig into BGP issues you'll see sometimes that traffic one direction takes a different route than traffic the other direction. I don't know of any simple and accurate way to learn the one way latency so I'm surprised they specified anything other than round trip time. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. That's possibly maybe true if someone's router or connection is overloaded and they are trying to make up for it with CoS policies while they save up for an upgrade. Otherwise it's an apology for a crappy network. That's the brutally honest truth. You can make a pretty good prediction with ping. sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation of voip traffic. let it run for awhile, then press ctrl+c and see how many packets were dropped and also check the mdev number. If mdev is low and packet loss is almost nothing then you can expect decent voice quality. It may not be a 100% perfect test, but I'll bet you a vast majority of the time I can do that test and tell you whether it's going to suck. latency by itself with low jitter and no packet loss just means delay. It's a matter of opinion and circumstance how tolerable delay is, but I think your 230ms ping is at the upper edge of what most people can live with. Much more than that and you'll be tempted to say 'over' at the end of sentence. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to find out one way latency
I would bet you get about the same result with the two providers.all else being equal. mdev (mean deviation) is a simple way to measure jitter, and you have to put in context with the min/avg/max numbers. If I had 7ms of deviation and average times of 4ms, that would be an issue because you would be likely to get packets out of order. But 7ms compared to 286ms probably means nothing. Your biggest problem with both providers is delay, but if you can tolerate the delay you have now, then you can probably tolerate the delay with the other provider. Also note that although packet loss is 0%, some packets are still dropped in both cases. One dropped packet means a small amount of audio is lost (depends on codec, but often 20ms). If those handful of dropped packets are scattered evenly then you wouldn't notice it, but it's common for them to occur in a cluster. If the 13 packets dropped in the first example all happened at once you would have lost 260ms of audioand you would certainly hear that. You may be able to tell by watching the periods appear on the screen when you run the ping command. Each period is a dropped packetif they accumulate in a burst then something is happening that you would hear on the phone. WOW.. That is the most complicated Ping I have ever seen.. :) This is the result I got. # ping -f -i .02 -s 180 -Q 0xb8 xx.xx.xx.xx /PING xx.xx.xx.xx (xx.xx.xx.xx) 180(208) bytes of data. . --- xx.xx.xx.xx ping statistics --- 15338 packets transmitted, 15325 received, 0% packet loss, time 352748ms rtt min/avg/max/mdev = 276.499/286.185/310.118/7.248 ms, pipe 15, ipg/ewma 22.999/284.882 ms / The same test with my Present SIP Provider gave me the result below. /10926 packets transmitted, 10913 received, 0% packet loss, time 244048ms rtt min/avg/max/mdev = 289.514/292.668/316.350/2.336 ms, pipe 15, ipg/ewma 22.338/292.941 ms / I suppose the value of mdev is much higher in the first case but 0% packet loss in both the cases. Does this mean that the voice quality is going to be real bad?? Thanks, Najim On Thu, Dec 1, 2011 at 6:33 AM, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. I am probably splitting hairs, but that's not always true because there's no guarantee that the reply traveled the same path as the echo request. If you dig into BGP issues you'll see sometimes that traffic one direction takes a different route than traffic the other direction. I don't know of any simple and accurate way to learn the one way latency so I'm surprised they specified anything other than round trip time. 'Ping time' is not an accurate predictor of SIP quality. A 'ping' is an ICMP Echo/reply packet and some routers consider them less important than 'data' packets and service them on an 'as resources permit' basis. That's possibly maybe true if someone's router or connection is overloaded and they are trying to make up for it with CoS policies while they save up for an upgrade. Otherwise it's an apology for a crappy network. That's the brutally honest truth. You can make a pretty good prediction with ping. sudo ping -f -i .02 -s 180 -Q 0xb8 [ip] gives a tolerable simulation of voip traffic. let it run for awhile, then press ctrl+c and see how many packets were dropped and also check the mdev number. If mdev is low and packet loss is almost nothing then you can expect decent voice quality. It may not be a 100% perfect test, but I'll bet you a vast majority of the time I can do that test and tell you whether it's going to suck. latency by itself with low jitter and no packet loss just means delay. It's a matter of opinion and circumstance how tolerable delay is, but I think your 230ms ping is at the upper edge of what most people can live with. Much more than that and you'll be tempted to say 'over' at the end of sentence. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. I just installed 3 Trixbox systems in KVM on Ubuntu. They're emergency PBX's for a few companies who lost their phone systems in a flood. They'll become real machines located on the customer premesis in the near future, but they've been running fine for a couple of weeks as virtual machines. One customer reported gaps in the hold music, but that was the only issue and I have no reason to suspect it's related to being virtual machine. I have not tried VirtualBox. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk hardware
Is there any reason not to run Asterisk on an Intel Atom board? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Force a SIP friend to use a certain IP?
Suppose I have two IP aliases on one asterisk box. I have to talk to SIP friend A using IP x.x.x.x and I have to talk to SIP friend B using IP y.y.y.y. (In case you're wondering, the reason is that we have two accounts with a service provider and different features and rates are tied to the two different accounts.) So I was hoping I would be able to set the source IP that we use when talking to the two different SIP friends. I see externip in general options, but is there nothing equivalent that can be set per user/peer? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force a SIP friend to use a certain IP?
So I was hoping I would be able to set the source IP that we use when talking to the two different SIP friends. I see externip in general options, but is there nothing equivalent that can be set per user/peer? Hi, as far as I know, you cant do this on a per peer basis. I suppose you run two asterisk daemons, each one of them on a different external IP. In this setup you can route calls from A over one asterisk daemon and calls from B over the other asterisk daemon. Sounds a little bit like an overkill scenario, but it woul work. best regards, Ruben Thanks, I was afraid of that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax from FXS to PRI
If I have a 4 port Digium FXS card and a single port PRI card on the same asterisk box, is it expected that I'd be able to plug a fax machine into the analog FXS port and have no problems sending or receiving faxes? Our connection to the Telco is on the PRI obviously. I don't recall the specific card models that we have, but I can check if it matters. Does the version of asterisk or Zaptel matter? My related question is this: In the scenario described above does the audio pass directly from one card to the other through the PCI bus or does it have to somehow be processed by software? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to proceed next
Does anybody ever update the software on the Panasonic phone system they had installed 30 years ago? Maybe if it ain't broke don't fix it. Hello list, I presently use the 1.4 releases because I enjoy sleeping at night. I understand that 1.4 reaches end-of-life in a little over 8 months (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions). I also know (as best as I can) that no genie is going to make Asterisk 1.4 go poof on this date. My clients would probably sleep better thinking they were running a PBX that didn't have this drop dead date however. Since 1.6.X has the same time constraints as 1.4, it seems it would be a waste of time going that direction. Should I go down the 1.8 .X path to have 4 years of time, but the headaches that have been documented here, or pursue the 10.X which is presently considered Beta? (is it really beta, or just relabeled 1.8?). Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10.0.0 better than 2.0.0?
So next is version 11 and then version 100? it has been mentioned that 10 is of course 2 ... think not in base 10 On 22 July 2011 22:26, Matthew J. Rothmr...@imminc.com wrote: Kevin P. Fleming: The versions all go to ten. Look, right across the board, ten, ten, ten and... Asterisk Users: Oh, I see. And most open source projects upgrade to two? Kevin P. Fleming: Exactly. Asterisk Users: Does that mean it's better? Is it any better? Kevin P. Fleming: Well, it's eight better, isn't it? It's not two. You see, most blokes, you know, will be running at two. You're on two here, all the way up, all the way up, all the way up, you're on two on your software. Where can you go from there? Where? Asterisk Users: I don't know. Kevin P. Fleming: Nowhere. Exactly. What we do is, if we need that extra push over the cliff, you know what we do? Asterisk Users: Put it up to ten. Kevin P. Fleming: Ten. Exactly. Eight better. Asterisk Users: Why don't you just make two better and make two be the top number and make that a little better? Kevin P. Fleming: [pause] Asterisk goes to ten. -- Sorry, couldn't resist. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom ip320 dtmf issues
Which dtmf method? I think we use inband here without issue. I am dtmf recognition issues. Out bound calls go though dahdi trunk/sangoma a400. Dtmf tones are not being recognized. Is there any issues with the latest polycom firmware? Sent from my android device. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk T38
In the simplest terms I can think of, I'm going to describe what I want to do and I want to know if it's possible in the current version of asterisk. Can I take a T38 call from an ATA, convert that back to analog and have asterisk screech that out on a POTS line to a remote fax machine. Would it work? And could I receive an incoming fax the same way? Please don't talk to me about alternatives to faxing. I can't take the fax machine away from the end user, they don't want to hear about it. I either need to make it work or tell them to get a POTS line. attachment: t38 diagram.jpg-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk T38
That's probably what I'm going to have to do. Thanks. I suppose that merely removing ATA and asterisk from the middle, and plugging a pots line into a fax machine is out of the question. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Level3 reseller needed
I'm in the Northeast US and looking for any recommendations on Level3 resellers. I don't do enough volume to go to Level3 directly. If there's anybody you'd definitely avoid I'd love to hear about that too. Thanks, Adam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Party ID issue
DON'T reply to people off list. And stop bloody top posting. Steve Is bottom posting your personal preference or is that a rule on this list? I have personally always found top posting easier to follow because the newer content is at the top. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Carrier needs more call examples
Ok list users, this is a question born out of curiosity, but if I'm having an intermittent problem and the carrier wants some examples of calls where the problem happened, what can they actually do with that information? I guess my implementation is relatively simple here and all I've got to look at is CDR's and the asterisk log. Does my carrier have tremendously more information they can look at? I ask because I gave them 2 or 3 examples and they want more, and I don't know what difference it makes whether they have 3 examples to look at versus 300. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Skip the whole NAT scenario. Put up an asterisk box with two network interfaces. One interface connects to the real world on your new IP address from your new ISP. The other interface can be on the same subnet as the windows box that you can't change. Set up a SIP trunk to your Windows box. Use packet 2 packet bridging in asterisk. Now that the emergency is over you can migrate off of your Windows thing at a more comfortable pace. You will be using someone else's public IP privately for awhile, but the main thing affected by that is your asterisk box won't be able to talk to anybody in that subnet in the outside world. You'll have to determine how bad of a thing that would be. BTW: What the heck is this software? Sounds like whoever wrote that wasn't thinking ahead. Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux/Asterisk on game consoles?
Out of curiosity why would you want to? Hello I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Normalize Voicemail Volume?
We generally get our voicemails emailed to us from asterisk, but some people's messages are extraordinarily loud or quiet. I don't suppose there is any feature to even out the volume level is there? ___ -- 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] Ghost ??
That would have to happen on the analog side of things, not within asterisk. If you have analog POTS lines I would talk to the telco about it. If you have analog phones connected to asterisk, then I'd wonder if there was anything near your wiring that might induce a signal into it. Or perhaps you have analog wireless phones? We are using asterisk and sometime when our guys are on call , they hear some voice of person and amazingly that person is NOT from our center. Any one faced this kind of thing ? ___ -- 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] voicemail number of rings
I'd be really happy if users could use the voicemail menu to change the number of rings until voicemail picks up. It seems like the current model of separate Dial and Voicemail commands would make that difficult, but is there any plan for such a feature in the future? How about a workaround or 3rd party add on? I store the dial timeout for each user in a database, so I know I could make my own little menu for them to set the number of seconds, but people are always a little stupefied by the fact that it's not on the voicemail menu. ___ -- 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] ATA recommendation??
Grandstream makes an 8-port unit which we've had success with, you could use three of them. Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can not control who is calling where... And the billing system will also be a problem in that case. Tnx... ___ -- 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] Area code 757 Car warranty calls
Verizon wireless filed a lawsuit against the perpetrators of the car warranty scam. I hope to hell they win. http://www.foxnews.com/story/0,2933,501404,00.html Cary Fitch wrote: The problem has two prongs - first we are in control of our own landlines and can use asterisk to screen whatever crap we wish before disturbing a real user or allowing a vm to get stored (but it would be nice not to have to). The other issue is we are not for the most part in any kind of control situation of our cellphones, and there is no way to stop that ring from happening and once it does it either needs to be answered or a vm dealt with. This is where the bigger players need to start living up to their responsibilities and not just ignore the problem. Well it will get me off my rant in this forum. Isn't that worth something? Seriously, as users some of us have one 2 line system and others are running multiple systems, absorbing hundreds of thousands of calls a day. Where the %#! warranty calls are coming from or not coming from is useful info. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, March 20, 2009 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Area code 757 Car warranty calls This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 ___ -- 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] mini-PCI FXS card?
Is there any product that's a single port mini-PCI FXS card? I'm aware of the Openvox A400M http://www.openvox.com.cn/products.php?genre_id=39, but I really only wanted one port. How about a single or dual port PCI or PCI express FXS card? Basically I wanted to build a small linux router with one or two phone ports. Alternatively, is there already a router or single board computer with FXS ports that I could run linux/asterisk on? ___ -- 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] mini-PCI FXS card?
Thanks Philipp, This and everything else I see out there is a bit more than I need :) I'm sure a single or dual port analog FXS card is not something most people want though, otherwise somebody would be selling it. Thanks anyway though. I'd recommend Sangoma's new B700 FlexBRI hybrid card (4 BRI ports, 2 FXS/FXO) http://www.sangoma.com/products_and_solutions/hardware/digital_analog_hybrids/flex_bri.html or the B600 (4 FXO, 1 FXS) http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html although that might be a bit mor than you need. Philipp Kempgen ___ -- 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] ATA gateway with 2 ethernet interfaces
I don't know of any ATA like that except the grandstream. The service provider grade way to do this would probably be a Cisco (or similar) with a T1 interface and a channel bank to break the T1 into 24 FXS ports. Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm out of business (at least until I replace the switch but that's usually too long for a busy system). The GXW4008 device is very useful for this scenario. It has 2 RJ45 ports (called WAN and LAN) and I've set them up in two local subnets. Not only does the ATA keep working without human intervention if one of the switches goes down but if both switches are up it can load balance between the two (simply by using DNS SRV with the same weights). Unfortunately, Grandstream in general doesn't seem to be very reliable although the latest GXW4008 firmware has proven to be quite stable in my case (previous releases were buggy). So I'm looking for alternatives to the GXW4008, even if it has to cost me more money. Does anyone know of an 8+ FXS ATA brand/model with 2 ethernet interfaces? Thanks in advance. Vieri ___ -- 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] Dailplan code for holiday detection?
I'm making extensive use of the MYSQL command.do you know if this behavior is considered a bug or not? This dialplan is illustrative of the particular problem of the MYSQL command in that no cleanup is performed if the dialplan terminates abnormally. If a device hangup occurs between the Connect and Disconnect, or worse, between the Query and the Clear, then extra resources will be consumed until a restart is performed. To avoid this problem, you should ensure that you always clear your query resources and disconnect your handles in the h extension. Or use func_odbc, which performs this sort of cleanup for you. ___ -- 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] Dailplan code for holiday detection?
Next question: When you say extra resources will be consumed until a restart is performed. Do you mean I have to restart asterisk to free up said resources? Will a reload do it also? This dialplan is illustrative of the particular problem of the MYSQL command in that no cleanup is performed if the dialplan terminates abnormally. If a device hangup occurs between the Connect and Disconnect, or worse, between the Query and the Clear, then extra resources will be consumed until a restart is performed. To avoid this problem, you should ensure that you always clear your query resources and disconnect your handles in the h extension. Or use func_odbc, which performs this sort of cleanup for you. ___ -- 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] Other lists
Does anybody know of a mailing list devoted to SIP device or ATA issues? This is a pretty high traffic list and I'd like to not clutter any more than I have to. Is there a polycom list for example? ___ -- 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] LNP Problems
Thanks everyone for the input. A CSR is nothing more than a listing of the numbers by your current provider on some sort of letterhead to indicate you actually are the subscriber who these numbers belong to (ie, you pay the bill for them). Is it necessary for the actual LNP process - no, not technically but companies require it to make sure they are not porting some else's numbers. Most CLEC's will just use a copy of your bill as the CSR. RBOC's have a more formal record which lists USOCs and other data that is completely unnecessary. The company doing the LNP will also need an LOA from you to request the CSR from the current provider. Time Warner most likely does have to give you one if they operate as a CLEC in your state or residence but it wont be you they give it to. You should provide TWTelecom with an LOA and then they can request the CSR from Time Warner. If the numbers in question are not numbers native to Time Warner - ie, Time Warner ported them from Bell or your regional LEC, then TWTelecom can force the issue and just port them by updating the LNP database with their service provider id and other appropriate information. Time Warner does not have to release the number to them for this. Regardless, its useless for you to bother calling Time Warner and ask for a CSR because the only people who would know what you are talking about are in the LNP/Carrier division and unless you work for another carrier, you wont get to them. Your new Telco will have to do this. If they cant accomodate this, I would find another provider. On Tue, Aug 12, 2008 at 3:42 PM, Adam Moffett [EMAIL PROTECTED] wrote: What is the deal with CSR's? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for one) are telling me that's wrong, that I don't need one and they don't have one to give me. Does anybody know what to do at this point? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LNP Problems
What is the deal with CSR's? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for one) are telling me that's wrong, that I don't need one and they don't have one to give me. Does anybody know what to do at this point? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 501 transfer feature
I can't transfer calls with my polycom 501's. Do I need to set up something in particular in the asterisk dialplan to make the feature work? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 transfer feature
Thanks for responding Kate. I do have a transfer button on the phone, and I follow the transfer process as described in the user's guide. When I press transfer the first caller is placed on hold and then I call the party I want to transfer to. At this point I'm supposed to press transfer again to connect the two parties together. Instead absolutely nothing happens, I can still press cancel to return to the first caller, but that's it. We have 3 of these phones and it used to work on all 3 of them. At some point we noticed it wasn't working any more on any of them but I'm not sure what changed. Any ideas? Thanks, Adam I think it should work standard (i.e. no special setup) Do you have a transfer button on the phone? Kate Adam Moffett wrote: I can't transfer calls with my polycom 501's. Do I need to set up something in particular in the asterisk dialplan to make the feature work? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any SLA alternatives?
The simplest way to do this is to use a different prefix for dialling out.
For example, if they dial out with a prefix of 9, use the shared number, and
if they dial out with a prefix of 8, use the private number.
Took the words right out of my mouth.
Basically if you have something like this:
exten = 1NXXNXX,1,Dial(outgoingtrunk/${EXTEN})
Add something like this:
exten = 91NXXNXX,1,Set(CALLERID(number)=sharedoutgoingnumber)
exten = 91NXXNXX,2,Dial(outgoingtrunk/${EXTEN})
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Re: [asterisk-users] New York Asterisk Users
Do you mean the city or the state of New York? I'm in NY, but a long ass way from NYC. This is an email to all* New York* based Asterisk users. For some time it’s been bugging me that we don’t have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I’m going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many Asterisk users there are in NY we’ll work out what to do from there. Cheers, Dean ___ -- 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 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 269.24.0/1462 - Release Date: 5/23/2008 7:20 AM ___ -- 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] Can I roll my own E911?
Assuming I only operate in one municipality (I do), and assuming I made some sort of connection to the emergency services center in this area, via SIP or a T1 or whatever, does asterisk have a way for me to send the E911 address 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] E911 Recommendations?
I'm in the same boat. And we don't need any snide comments because this is a potential liability. Municipalities don't provide E911, they are users of E911 data. If you are not a phone company and you want the E911 data updated with correct addresses, then you need to pay someone to do that for you. That is unless I grossly misunderstand it. On Mon, Apr 14, 2008 at 02:05:18PM -0500, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? Wow. E911 providers are *municipalities*, aren't they? :-) Could you vague that up a bit, Doug? (Or should I be able to generalize that phrasing into what you actually mean, if I expect to get along here? :-) Cheers, -- jra ___ -- 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] E911 Recommendations?
Ok so did anybody have recommendations? How's 911Enable.com? Anybody have recommendations for a reliable, good valued, E911 provider? ___ -- 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] is this possible..
I think he's talking about an automated system. It's definitely possible with asterisk, whether or not it's a good idea. I really see this is useless since we alreadu got pricegrabbers buy.com and froogle they all list the itme in stock on the site there is really no need for a $30k a year operator to read it for the person. just my $0.02 On 3/6/08, blackwater dev [EMAIL PROTECTED] wrote: I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls person B to see if the item is available, cost, etc. I'd like for the person searching to be able to click on 10 items they are interested in then click another button which would have asterisk start at the first, call person B, ask if the item is available, if yes, then call person A and connect the two, if not, it says thanks, and calls the next person on the list. Is this possible with Asterisk? Second, anyone looking for some contract work to help get this prototype running? Thanks! ___ -- 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] PPP dialout via * server
I've been able to run low speed modems through a SIP ATA and an IAX trunkbridged by asterisk. I would assume that in this day and age any modem application through asterisk is probably for either a remote console or some sort of control system. Either way using SIP and a very low speed should work for that type of stuff. ___ -- 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] ata device but for a soundcard
So you want a device that will answer a SIP call, and play the audio out to a speaker? You're looking to build a PA system then? Get a regular ATA and plug something like this into it: http://www.vikingelectronics.com/products/view_product.php?pid=199 I am looking for an ATA like device but instead of VOIP to analog phone I want VOIP to low level audio out. Something that looks like a sound card output. I know I can use cheap PC's but that then you have HD's to setup etc... HD failures etc... Anyone know of something like that? Jerry ___ -- 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] Best ATA. Period.
Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works? ___ -- 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] Call drops to fast busy
Ok. The problem that prompted my best ata question is this: I have a person connecting to our asterisk box remotely with a generic ATA. It was actually purchased from Tiger Netcom and is based on an HTTEL chipset. This person says that sometimes they will be in the middle of a call and it will drop and go straight to a fast busy. This is not something I've encountered with anyone else. So is this a: 1) Network problem 2) Asterisk problem 3) Defective ATA 4) ___ -- 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] Best ATA. Period.
H. Does Digium make a card for that? Tin cans and string. Very easy to set up. Very easy to diagnose if it does not work (check for tear in brown paper diaphragm or string not tight). All other devices are subject to failure and counting on anything to just work is a short path to frustration and failure. ___ -- 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] Best ATA. Period.
In all seriousness, my requirements were a little silly. A Cisco router can fail just as a netgear router can. But I think we would find Cisco failures to be statistically less likely. I also think we can agree that not all devices of a certain type are created equal. Do you have any opinions on which VoIP products are more likely to be consistent and reliable? Tin cans and string. Very easy to set up. Very easy to diagnose if it does not work (check for tear in brown paper diaphragm or string not tight). All other devices are subject to failure and counting on anything to just work is a short path to frustration and failure. ___ -- 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] IAXy ringing
When I make calls from my IAXy I don't hear any ringing most of the time. I've tried using the r option on the asterisk dial application to indicate ringing to the calling party but that didn't make a difference. Anything else I can try?___ -- 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] CallerID Number incorrect in SIP packet
in sip.conf under the definition for the sip user add callerid=whatever - Original Message - From: Lutgring, Sam To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, January 08, 2008 4:37 PM Subject: [asterisk-users] CallerID Number incorrect in SIP packet I am having an issue with the CallerID Number not being passed to my phone in the SIP packet. The CallerID Name is passed just fine and displayed on the phone with no issue. I have done a NoOp() in my extension.conf and successfully seen both the CallerID name and number correctly. So that leads me to believe that Asterisk is handeling it correctly. However, when I do a packet capture of the SIP packet sent from the Asterisk server to the phone, I do not see the CallerID Number but instead see the registered user name of the phone: The lutgrins-G-2433 is the user name that my phone is registered as. I would expect to see sip:[EMAIL PROTECTED] instead of what I am seeing. Both the phone and the server are running on the same network segment (no NAT involved). Any help would be appreciated. I am running Asterisk version 1.4.11 -- ___ -- 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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1213 - Release Date: 1/7/2008 9:14 AM Outlook.jpg___ -- 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] With rtcachefriends=yes, when do realtime changes take effect?
So I guess the answer is that updates to existing peers never happen until you make them happen. I really think an update on reregistering would be better, then I can just tell the user to reset their phone. Or a periodic automatic pruning. For someone who doesn't know any C and therefore can't write a patch.is there any right way to make a feature request? Or you can prune the specific user entry and it will look it up again. Anthony Francis wrote: Adam Moffett wrote: I asked this question last week and never got an answer. I also didn't find the answer in the wiki. I think it would be nice if asterisk would check the database again if the user re-registers, but it doesn't seem to do that. A periodic update would be ok too, but it doesn't seem to do that either. It seems like changes never happen until a reload.if that is the case then doesn't rtcachefriends completely defeat the purpose of realtime SIP 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 New entries take effect immediately, however changes require a sip reload. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1207 - Release Date: 1/2/2008 11:29 AM ___ --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] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
Thanks nice script.
But why au files in addition to gsm?
- Original Message -
From: dave cantera
To: Asterisk Users Mailing List - Non-Commercial Discussion ; [EMAIL
PROTECTED]
Sent: Tuesday, January 01, 2008 11:27 AM
Subject: Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the current
directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
#convert-all.sh
#
#convert all *.wav files to .gsm .au formats
#
if [ null${1} == null ]
then
FILE_LIST=`ls *.wav`
else
FILE_LIST=`ls ${1}*.wav`
fi
LOG=./log_convert.log
echo === ${LOG}
echo started at `date` ${LOG}
echo Removing all current .gsm files...
rm -f *.gsm
for FNAME in ${FILE_LIST}
do
echo --- -
echo ${LOG}
echo Processing ${FNAME}...
echo Processing ${FNAME}... ${LOG}
BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`
echo making ${BASEFNAME}.gsm...
echo making ${BASEFNAME}.gsm... ${LOG}
#sox -q -V -c 1 ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample -ql
2${LOG}
sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql 2${LOG}
echo ${LOG}
echo making ${BASEFNAME}.au...
echo making ${BASEFNAME}.au... ${LOG}
sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample -ql
2${LOG}
done
Vincent wrote:
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
... I recorded a sample of my voice using XP's Sound Recorder, then
ran the following :
sox test_wav.wav -r 8000 -c 1 -s -w test_wav_out.wav resample -ql
But it seems like I'm missing the codec or something:
===
-- Executing [EMAIL PROTECTED]:2] Playback(SIP/2000-0871d000,
/usr/local/lib/asterisk/test_wav_out.wav) in new stack
WARNING[37390]: file.c:563 ast_openstream_full: File
/usr/local/lib/asterisk/test_wav_out.wav does not exist in any format
WARNING[37390]: file.c:866 ast_streamfile: Unable to open
/usr/local/lib/asterisk/test_wav_out.wav (format 0x4 (ulaw)): No such
file or directory
===
Here's what core show file formats says:
===
Format Name Extensions
gsmwav49 WAV|wav49
slin wavwav
adpcm voxvox
slin slnsln|raw
g722 g722 g722
ulaw au au
alaw alaw alaw|al
ulaw pcmpcm|ulaw|ul|mu
ilbc iLBC ilbc
h264 h264 h264
h263 h263 h263
gsmgsmgsm
g729 g729 g729
g726 g726-16g726-16
g726 g726-24g726-24
g726 g726-32g726-32
g726 g726-40g726-40
g723 g723sf g723|g723sf
18 file formats registered.
===
Am I missing something in the configuration files, or maybe I'm
missing some module?
Thank you.
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I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
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856.380.0894
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Version: 7.5.516 / Virus Database: 269.17.13/1205 - Release Date: 12/31/2007
3:32 PM
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[asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?
I asked this question last week and never got an answer. I also didn't find the answer in the wiki. I think it would be nice if asterisk would check the database again if the user re-registers, but it doesn't seem to do that. A periodic update would be ok too, but it doesn't seem to do that either. It seems like changes never happen until a reload.if that is the case then doesn't rtcachefriends completely defeat the purpose of realtime SIP 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] T.38 fax solution, opinions?
If you mean faxing in audio it's hit or miss. We do it here and maybe have an error every 6 pages or so. I wouldn't sell it to a customer as a solution. How about fax machines talking directly to spa2102 and then out the pri or am I missing something? ___ --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] Only call me once
[EMAIL PROTECTED] wrote: Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected. ___ --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 Yup store the number in DB if its new, if it's already in db reject. This is an incredibly simple thing to do, you can even just use the Asterisk internal DB for simplicity. Anthony Yeah fairly easy, but why would you want to? Is this part of a verification process like throwaway URL's that get emailed to me when I sign up at a web site? ___ --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] Newb Question
I'm pretty sure asterisk won't do that without modification. You'll need to do packet sniffing and decode the datathere may be products that do this, but asterisk is not it. And we're assuming the calls are unencrypted? I inherited an office with phones that are hosted off-site. Everything is skinny and G729. I see that the FreeBSD asterisk port comes with a G729 codec. I want to record everything. If I use port mirroring on my switch, is it possible to configure asterisk to record and assemble packets that it doesn't otherwise route? Is it insane to user asterisk for this purpose? Advice or a link to a howto would be greatly appreciated. ___ --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] Correct syntax for IF()?
A simpler example reveals the problem:
exten = 188,1,Noop(${STAT(e,/bin/ls)})
exten = 188,2,Noop(${STAT(e,/not/there)})
Try that and you'll find that STAT(e,/whatever) returns 1 if the file is
found and NOTHING if the file is not found.
This method should work:
${IF($[${STAT(e,/tmp/${CALLTIME}.wav)} = 1]?${CALLTIME}.wav:)}
So if the file exists you'll have
1 = 1 (true)
if not you'll have
= 1 (false)
Keep in mind that quotations marks don't mean anything special to
asterisk. It's just another character, and here they're just used to
enclose the empty string that is returned by STAT. We could probably
use any other character for this purpose, but quotes probably seem more
natural to us English speakers.
crybaby mode
This bizarre syntax is the thing I hate most about asterisk. For people
like you and me with some scripting or programming experience it would
seem natural to use IF the way you wanted to, but asterisk is just weird
in this way.
/crybaby mode
On Mon, 26 Nov 2007 23:40:37 +0100, Turbo Fredriksson
[EMAIL PROTECTED] wrote:
What you do is you always write the beginning _and_ the end at once. Never
try to do them
'later'...
Thanks guys. I think I found where it goes wrong:
==
1. /tmp/test.wav exists - the $[] is true:
exten = h,n,Set(CALLTIME=test)
exten =
h,n,Set(WAV_FILE=${IF($[${STAT(e,/tmp/${CALLTIME}.wav)}]?${CALLTIME}.wav:)})
exten = h,n,Verbose(WAV_FILE is ${WAV_FILE})
==
2. /tmp/dummy.wav doesn't exist - the $[] is false:
exten = h,n,Set(CALLTIME=dummy)
exten =
h,n,Set(WAV_FILE=${IF($[${STAT(e,/tmp/${CALLTIME}.wav)}]?${CALLTIME}.wav:)})
exten = h,n,Verbose(WAV_FILE is ${WAV_FILE})
= WARNING[5296]: func_logic.c:107 acf_if: Syntax
IF(expr?[true][:false])
==
For the false part, I tried the following, none works:
- nothing
- :
- :
Is it a known bug, and does Asterisk 1.4.14 solve this?
Thank you.
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Re: [asterisk-users] asterisk web interface
So what you actually want a web based phone? Jody Gugelhupf wrote: hi there :) i want to have a website, which offer the following: # Integrated Web Dialer (Click-to-Dial) Easy to make your own. The only question is "integrate into what?" i don't know, maybe easy for you but not for me ;) just a webinterface in php or twisted.web maybe? # Workgroup Answering Machine # Monitor recent call history (CDR) # Listen/manage voicemail # Monitor incoming calls and call history of incommng calls Debian has freePBX and DeStar packages. I think both provide most or all of what you need. If not, what do you find missing? well as i said i tried them out, but i did not find or only partial but not all together, the following features for sip only: webdialer, listen/manage voicemail, Monitor incoming calls and call history of incommng calls with call back function, pick-up incoming calls through the webinterface, online phonebook these functions integrated into a simple plain webinterface would be great, so anyone knows one? thx jody :D Ask a question on any topic and get answers from real people. Go to Yahoo! Answers and share what you know at http://ca.answers.yahoo.com ___ --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] VoiceMail Access
I created a *9 extension which executes VoiceMailMain with the callerid number as the argument. Then of course the voicemail box just has to be the same as the phone number. Then we just have another DID for outside access. * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Mike Hammett wrote: I was looking at the ILECs' web sites to determine how their users access voicemail. I looked at ATT, Verizon, Qwest, and Embarq. They supported one or a combination of the following for calling from your phone: *98 #55 Toll free number Your number A varying phone number, based on your number's location. Calling from anywhere else, they supported: Hitting star when you hear your greeting when calling yourself Toll free number What method should I use for my users checking their voicemail? Can Asterisk voicemail be made to accept hitting * during the greeting to enter the voicemail system? If they call their own number, how do I get Asterisk to recognize that and take them to the voicemail system? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --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] Video Door Phone
You could probably make something work, but instead of trying to pound a nail with a wrench.buy a door entry control system. * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Smith, Rick wrote: I have a customer that has a campground. Wants to see who's at the gate, remotely, via camera, and talk to that person through a "traditional squawk box" and be able to open the gate remotely from that phone. He doesn't want to have a separate camera feed, etc, he wants to do it all on one phone. Does such a way to do this exist by using Asterisk and some kind of relay system / Video phone ? R ___ --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] select menu
My suggestion:
[your incoming context]
#answer the phone
exten = s,1,Answer()
#playback recording but also accept extensions
exten = s,2,Background(your_gsm_recording)
#wait for caller to dial extension
exten = s,3,WaitExten(10)
#if they haven't hit an extension yet, play the message again
exten = s,4,Background(your_gsm_recording)
#give them one more chance
exten = s,5,WaitExten(10)
#send them to a default extension...maybe they have rotary phone
exten = s,6,Dial(SIP/101|30|tm)
#if all else fails, hangup
exten = s,7,Hangup()
# dynamic extension which makes 1=101, 2=102, etc.
exten = _X,1,Dial(SIP/10${EXTEN}|30|tm)
*
Adam Moffett
Plexicomm, LLC
[EMAIL PROTECTED]
ph: 866-759-4678x104
*
Josu Lazkano Lete wrote:
Hello everybody.
I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3).
if he choose 1 it will redirect to 101 extension
if he choose 2 it will redirect to 102 extension
if he choose 3 it will redirect to 103 extension
my extensions.conf is this one:
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
sorry about my english,
thanks to all
be
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Re: [asterisk-users] SIP Problems continue...
I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/0 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this ...the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * ___ --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] Mobile Number to Mobile carrier mapping
Try this: http://puck.nether.net/npa-nxx/ * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Ritesh Agrawal wrote: Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh ___ --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] 1.4 memory leak?
Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this afternoon I'm up to 11megs: VIRT RES SHR SWAP CODE DATA 30932 11m 560818m 1012 17m Is this a real issue or do I have something else going on? -- * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * ___ --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] 1.4 memory leak?
You're right, 11megs isn't scary at all. It's the 106 megs from Monday that worried me. * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Steve Finkelstein wrote: With all due respect, I believe you might be a bit paranoid. 10-11M is quite normal for the linux kernel to allocate for asterisk. It's not necessarily what the process is using, but that's just how memory management works within the kernel. What's 10-11M of RAM these days anyway? - sf Adam Moffett wrote: Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this afternoon I'm up to 11megs: VIRT RES SHR SWAP CODE DATA 30932 11m 560818m 1012 17m Is this a real issue or do I have something else going on? ___ --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] 1.4 memory leak?
Also the system I upgraded to 1.4.4 hasn't grown past 11 megs. Another system (with less usage!) still running 1.4.0 has gotten up to nearly 20 in the same time period. This is RES reported by top and ps btw. I understand that VIRT is not real usage, but my understanding was that RES was actual usage. Is that not the case? And do you think 106 megs is normal for a system that only handles 2-3 simultaneous calls? * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Steve Finkelstein wrote: With all due respect, I believe you might be a bit paranoid. 10-11M is quite normal for the linux kernel to allocate for asterisk. It's not necessarily what the process is using, but that's just how memory management works within the kernel. What's 10-11M of RAM these days anyway? - sf Adam Moffett wrote: Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this afternoon I'm up to 11megs: VIRT RES SHR SWAP CODE DATA 30932 11m 560818m 1012 17m Is this a real issue or do I have something else going on? ___ --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] what are these and can they be fixed?
I can't tell you exactly what it means, but you can make it go away by not logging debug information. look in /etc/asterisk/logger.conf Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:26 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:26 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:27 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found ___ --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] call drops after one ring
I'm trying to make a call through 2 asterisk servers, and the call
simply hangs up after one ring.
The path is like this:
ATA1 - (SIP) - server1 - (IAX) - server2 - (SIP) - ATA2
The two ATAs are registered to their respective asterisk servers and can
make and recieve calls to local extensions and call out via IAX to our
provider. But for some reason I can't call one from the other.
all I have in IAX.conf in both cases is something like this:
[theOtherServer]
type=friend
context=default
allow=all
host=1.2.3.4
qualify=yes
in extensions.conf on server1 under the default context I have:
exten = 103,1,Dial(IAX2/theOtherServer/${EXTEN})
on server2 I have:
exten = 103,1,Dial(SIP/ATA2)
when I call 103 from ATA1, ATA2 rings exactly one time and the call
disconnects. Is there something additional I need to make this work?
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Re: [Asterisk-Users] hunt groups
What I would like to do is… exten = 1000,1,Dial(sip/1000)(zap/g1,97837560) exten= 1000,2,Voicemail(u1000) Basically a follow me app that rings numerous interfaces and allows me to answer or it to time out and go to vmail. I didn’t include the time out here as I am hoping someone can tell me where that needs to be. I really don’t want to make the caller ring one interface and then the other. Ideally I would be able to press pound after answering so that it didn’t continue to ring the other interface. Most of the apps that I saw do this are basically the same as forwarding the extension, any system can do that and I know asterisk is better than that. Either put the Dial commands in sequence with a short timeout, or put multiple arguments to the dial command separated by Option 1) exten = 1000,1,Dial(SIP/1000|15) exten = 1000,2,Dial(Zap/g1,97837560|15) rings each extension for 15 seconds option 2) exten = 1000,1,Dial(SIP/1000Zip/g1,97837560) rings both extensions at oncefirst one to answer is the winner. ___ --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] embedded hardware for Asterisk?
Hi, Is there any specific made embedded hardware designed VoIP software or Asterisk? I want to build a router that have VoIP enabled, so that I can use it connect to a VoIP ISP. Thanks Sam There are plenty of small form factor boards you could start with (http://www.linuxdevices.com/), but I assume you'll want something with one or more analog telephone ports. I would also be interested if anyone knows of any hardware that's small, affordable, can run linux, and can have FXO/FXS ports. Thanks, Adam ___ --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 Jitter Buffer for 1.2.5
jitterbufferfor svn trunk + jitterbuffer jitterbuffer-1.2 for 1.2 + jitterbuffer test-this-branchfor the test branch with a lot of cool stuff including the jitterbuffer I installed the jitterbuffer-1.2 branch and I have a few questions. First and foremost I'm getting hundreds of lines like this in my log file: Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 The console shows something similar: Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:10 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 My log file is going to be very big today. What could be responsible for frames (every frame?) having invalid timing info? Second I'm not sure if it's actually doing anything. For testing, I tried setting the max size to 2000ms and implementation to fixed.if I'm reading the comments in the sample config correctly that should create a 2000ms fixed jitter buffer, which in turn should mean a 2 second delay in audio, but I wasn't hearing any delay at all. Is this not a valid way to test whether the jitter buffer is doing something? ThirdI'm interested in a way to create some jitter ;) I was thinking I might take an ethernet hub and try to saturate it with several simultaneous large file transfers or something like that. Another possibility might be an 802.11 wireless connection at a fairly long range. If anyone knows of a more convenient way for me to create a jittery connection I'd be very interested. ___ --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] impact of qualify=yes
Anyone have any information on the performance impact of using qualify=yes for hundreds (500ish) of SIP UAs? I have seen tidbits on qualifyspreading=yes, but not enough to understand what it does. I assume lessens the peak load of qualify sip options queries? Thx! Qualify=yes means we send one SIP packet to the sip user and receive one packet back, and calculate a round trip time. And I think this happens around once a minute. I can't imagine the performance impact being very big. The PC on my desk can do 2000 ICMP pings in 10 seconds with no impact whatsoever.qualifying SIP agents can't be much worse. But I am not an expert on the matter. ___ --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] Jitter buffer for SIP channels (OT?)
This might be a better question for the dev list, but I don't think they want to be bothered by my silly questions. Does anyone know when we can expect to see a jitter buffer for SIP channels? I know they've been working on a generic jitter buffer since around last summer, just wondering if there's been any progress. ___ --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] status on jitter buffer for SIP/RTP? (OT?)
This might be a better question for the dev list, but does anyone know the status of a jitter buffer for SIP channels? I know they created a generic jitter buffer and implemented it for IAX channels. Does it work yet for SIP? Like is it there and disabled or not there at all? ___ --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] Best ATA for general residential deployment??
Many thanks to everyone for their input. We have been using sipura 1001 and 2002 units and they work great as a SIP adapter, but something that can also function as a router would be more useful to us. Does anyone have any comments on the Sipura 2100? What about a battery backup? Time Warner cable in this area provides a cable modem + ATA device that includes a sealed lead acid battery inside. So in a power failure the customer's phone can still function. Is anyone aware of an Ethernet to Ethernet router + ATA that also has a battery backup. I realize a UPS would do the job, but it's overkill. Thanks again, Adam ___ --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] Configure DID
In the most basic case you create a SIP user and create extensions that point to those SIP users. in sip.conf: [sipuser1] username=sipuser1 secret=123456 type=friend host=dynamic disallow=all allow=ulaw (-- put your most preferred codec here) allow=gsm (-- other codecs you will support on subsequent lines) in extensions.conf: exten = 6071234567,1,Dial(SIP/sipuser1|60)(--- replace with your actual DID) I also suggest: exten = 16071234567,1,GoTo(6071234567|1) exten = 1234567,1,GoTo(6071234567|1) (-- these lines allow for the number to be dialed in different ways and still get to the SIP user) You could also create arbitrary extensions for your internal use: exten = 101,1,GoTo(6071234567|1) exten = 102,1,GoTo(some other extension) Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj. ___ --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] Best ATA for general residential deployment??
I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want to set it up and never look at it again ;) The way you make money on lower cost residential services is to make sure you spend as little labor as possible after the fact. If we have to install a $200 part, we'll make that money back with the monthly fee over time as long as we don't have to go back to, it or replace it, or spend a lot of time on the phone doing support. 2) Second priority is remote provisioninga truck roll to change configurations is not acceptable. A web or telnet interface is tolerable, but tftp or http auto configuration is desireable. 3) Third priority is pricefor obvious reasons Perhaps the biggest issue is we don't want to have to supply a router or switch in addition to the ATA. It's a lot of extra cabling that people might screw up, extra parts that might break, extra time for the installation, etc. Ideally, either a device that functions as an ethernet bridge (like vonage ATA's) so that it can be positioned in-line with other equipment; or a combination router/SIP adapter. The absolute best thing in the world might be a combination router, 802.11 AP, 4 port ethernet switch, and SIP adapter with a backup battery. Plug in one box and you're done. If the router can be reconfigured as a bridge (for customers who prefer their own router) so much the better. Any reccomendations would be welcome. ___ --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] strange voicemail issue
with this in extensions.conf: exten = xxx,x,voicemail([EMAIL PROTECTED]) I get this in the log: -- Executing VoiceMail(SIP/officeata1-5836, [EMAIL PROTECTED]) in new stack Jan 16 09:20:50 WARNING[2700]: app_voicemail.c:2379 leave_voicemail: No entry in voicemail config file for 'ales' no voicemail entry for ales? why is the first 's' chopped off? To make it more interesting, if I add the |s option thusly then everything works fine. exten = xxx,x,voicemail([EMAIL PROTECTED]|s) this is version 1.2.0 Anyone have any comments? ___ --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] IAX2-SIP dropped calls
Apparently we've been having calls sporadically drop. We're using an IAX outbound trunk and SIP adapters on the inside. Below is a log excerpt detailing one of the calls which dropped, and it looks largely normal to me except for this: Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging channels IAX2/teliax-2 and SIP/davidblanco-e02c Can missing one IAX frame result in a dropped call? Seems pretty fragile if that's the case. Would enabling the jitter buffer mitigate this? Any other suggestions? Jan 5 13:29:51 VERBOSE[29852] logger.c: -- Accepting UNAUTHENTICATED call from 208.139.204.245: requested format = ulaw, requested prefs = (g729|ulaw|g726|gsm), actual format = gsm, host prefs = (gsm|ulaw), priority = mine Jan 5 13:29:51 VERBOSE[3776] logger.c: -- Executing Dial(IAX2/teliax-2, SIP/davidblanco|30|tr) in new stack Jan 5 13:29:51 DEBUG[3776] chan_sip.c: Setting NAT on RTP to 524288 Jan 5 13:29:51 DEBUG[3776] chan_sip.c: Outgoing Call for davidblanco Jan 5 13:29:51 VERBOSE[3776] logger.c: -- Called davidblanco Jan 5 13:29:51 DEBUG[29854] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jan 5 13:29:51 DEBUG[29854] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jan 5 13:29:51 VERBOSE[3776] logger.c: -- SIP/davidblanco-e02c is ringing Jan 5 13:29:57 DEBUG[29854] chan_sip.c: Acked pending invite 102 Jan 5 13:29:57 DEBUG[29854] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Jan 5 13:29:57 VERBOSE[3776] logger.c: -- SIP/davidblanco-e02c answered IAX2/teliax-2 Jan 5 13:29:57 DEBUG[29852] chan_iax2.c: Ooh, voice format changed to 2 Jan 5 13:29:59 DEBUG[29852] chan_iax2.c: Peer lastms 70, historicms 70, maxms 2000 Jan 5 13:30:15 DEBUG[29852] chan_iax2.c: Peer lastms 28, historicms 28, maxms 2000 Jan 5 13:30:59 DEBUG[29852] chan_iax2.c: Peer lastms 71, historicms 71, maxms 2000 Jan 5 13:31:07 DEBUG[29852] chan_iax2.c: Immediately destroying 2, having received hangup Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging channels IAX2/teliax-2 and SIP/davidblanco-e02c Jan 5 13:31:07 DEBUG[3776] chan_sip.c: update_call_counter(davidblanco) - decrement call limit counter Jan 5 13:31:07 DEBUG[3776] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 5 13:31:07 VERBOSE[3776] logger.c: == Spawn extension (default, 6078210976, 1) exited non-zero on 'IAX2/teliax-2' Jan 5 13:31:07 DEBUG[3776] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record. ***CDR STUFF OMITTED*** Jan 5 13:31:07 DEBUG[3776] chan_iax2.c: We're hanging up IAX2/teliax-2 now... Jan 5 13:31:07 DEBUG[3776] chan_iax2.c: Really destroying IAX2/teliax-2 now... Jan 5 13:31:07 VERBOSE[3776] logger.c: -- Hungup 'IAX2/teliax-2' ___ --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] Announcing a call transfer
With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says so and so is on the phone for you, I say ok put him through, she hangs up and I am connected to the caller. With [EMAIL PROTECTED] I can it # then the extension to transfer to and it will ring there. But is there a simple way to announce the call before you transfer it. If not, does anyone have any good work arounds for this. There is a feature called attended transfer which does what you want. Receptionist dials the attended transfer code, followed by your extension. The caller hears hold music while the receptionist announces the call to you. When she hangs up you get the call. If you hang up before she does, the call goes back to her. It can be enabled in the features.conf file. Under the [featuremap] section add atxfer = code on my system it's atxfer =*2 so I dial *2 followed by the extension to do attended transfer. However, I don't know anything specific to [EMAIL PROTECTED], so if it's different than a stock asterisk setup then I don't 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
Re: [Asterisk-Users] Delays in IVR
;extensions for dan and adam ;dan - since people already know dan as extension 3, we keep that for compatibility exten = 3,1,GoTo(Pleximenu|103|1) exten = 103,1,GoTo(default|103|1) ;adam exten = 104,1,GoTo(default|104|1) The bottom of the dialplan is your culprit here. It's waiting the additional time because it's not sure whether or not you're going to enter 103 or 104 as opposed to just 1, so it's waiting for the digit timeout to be sure. Several people made that suggestion, but I had already tried it with those extensions commented out. Would anything be neccesary to make the change take effect aside from extensions reload? ___ --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] Delays in IVR
I set up an IVR awhile back. press 1 for sales, press 2 for support etc etc. Everything works fine except when you enter your option there is a 7 or 8 second pause before the next step is taken in the dial plan. I assume it's waiting to see if I'm going to dial more digits, but is there a way to reduce this delay? Thanks in advance. ___ --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] Delays in IVR
I set up an IVR awhile back. press 1 for sales, press 2 for support etc etc. Everything works fine except when you enter your option there is a 7 or 8 second pause before the next step is taken in the dial plan. I assume it's waiting to see if I'm going to dial more digits, but is there a way to reduce this delay? Yes, don't have overlapping extensions. i.e. either don't have an option 7 or 8 or don't number your extensions starting with 7 or 8 ___ I don't actually have an option 7 or 8.I was attempting to say that there is a 7-8 second pause after selecting your option. ___ --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] Delays in IVR
Please post the appropriate section in extensions.conf that is
responsible for the IVR's operation.
You asked for it.
The pleximenu context is reached from the default context by a simple
goto, as in:
exten = [ourphonenumber],1,GoTo(pleximenu|s|1)
Everything works as I expect it to except for the long delay between
dialing your option and actually getting your option.
[pleximenu]
exten = s,1,Answer()
exten = s,2,GoToIfTime(${BUSHOURS}?pleximenu|s-OPENHOURS|1)
exten = s,3,Noop(Must not be business hours)
exten = s,4,GoTo(pleximenu|s-OFFHOURS|1)
exten = s-OPENHOURS,1,Wait(1)
exten = s-OPENHOURS,2,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,3,WaitExten(15)
exten = s-OPENHOURS,4,Background(plexicomm/Main_Greeting)
exten = s-OPENHOURS,5,WaitExten(15)
exten = s-OPENHOURS,6,Hangup()
exten = s-OFFHOURS,1,Wait(1)
exten = s-OFFHOURS,2,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,3,WaitExten(15)
exten = s-OFFHOURS,4,BackGround(plexicomm/off_hours_greeting)
exten = s-OFFHOURS,5,WaitExten(15)
exten = s-OFFHOURS,6,Hangup()
;sales
exten = 1,1,Wait(1)
exten = 1,2,GoToIfTime(${BUSHOURS}?pleximenu|1-OPEN|1)
exten = 1,3,Noop(Must be off hours)
exten = 1,4,GoTo(pleximenu|1-OFFHOURS|1)
exten = 1-OPEN,1,Playback(plexicomm/hold_for_sales)
exten = 1-OPEN,2,Noop()
exten = 1-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 1-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 1-OPEN,5,Playback(plexicomm/sales_unavailable)
exten = 1-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 1-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 1-OPEN,8,Hangup()
exten = 1-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 1-OFFHOURS,2,Hangup()
;support
exten = 2,1,Wait(1)
exten = 2,2,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
exten = 2,3,Noop(Must be off hours)
exten = 2,4,GoTo(pleximenu|2-OFFHOURS|1)
exten = 2-OPEN,1,Playback(plexicomm/hold_for_support)
exten = 2-OPEN,2,Noop()
exten = 2-OPEN,3,Dial(${OFFICEPHONES}|30|m)
exten = 2-OPEN,4,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 2-OPEN,5,Playback(plexicomm/support_unavailable)
exten = 2-OPEN,6,Voicemail([EMAIL PROTECTED]|s)
exten = 2-OPEN,7,Playback(plexicomm/thanks_for_interest)
exten = 2-OPEN,8,Hangup()
exten = 2-OFFHOURS,1,voicemail([EMAIL PROTECTED])
exten = 2-OFFHOURS,2,Hangup()
;Starts a variable called ATTEMPT at 1
; tries calling ONCALLPHONES
; increments ATTEMPT variable by 1
; tries again until ATTEMPT = 4
; should be 3 attempts total
; set ONCALLTIMEOUT to a number of seconds before your voicemail
picks up.
exten = 9,1,GoToIfTime(${BUSHOURS}?pleximenu|2-OPEN|1)
;we shouldn't be doing this during business hours
exten = 9,2,Playback(plexicomm/page_support)
exten = 9,3,Set(ATTEMPT=1)
exten = 9,4,GoToIf($[${ATTEMPT} : 4]?9-FAILED|1)
exten = 9,5,Dial(${ONCALLPHONES}|${ONCALLTIMEOUT}|m)
exten = 9,6,Set(ATTEMPT=$[${ATTEMPT} + 1])
exten = 9,7,Playback(plexicomm/keep_paging)
exten = 9,8,Wait(2)
;waiting 2 seconds to allow cell connections to terminate
exten = 9,9,GoTo(pleximenu|9|4)
exten = 9,10,Hangup()
exten = 9-FAILED,1,GoTo(pleximenu|2-OPEN|5)
;extensions for dan and adam
;dan - since people already know dan as extension 3, we keep
that for compatibility
exten = 3,1,GoTo(Pleximenu|103|1)
exten = 103,1,GoTo(default|103|1)
;adam
exten = 104,1,GoTo(default|104|1)
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Re: [Asterisk-Users] ad hoc conferencing-reg
I've think I've been working on the same thing. Many SIP phones have a
built in conferencing feature...but they may not all work the same and
may have all different instructions. So doing it in asterisk is
preferable to me so I can give users one set of instructions for it.
It's not a simple straightforward thing like threewaycalling= on in
zapata.conf. For SIP you have to create an extension that executes a
macro which dynamically creates a meetme conference or adds a caller to
an existing one. Then you create an extension that goes to that macro.
Person A can then call person B, transfer person B to the conference
extension, call Person C, transfer Person C to the conference extension,
then call the conference extension to add themselves to the conference.
At least that's the ideaI haven't quite got it working perfectly ;)
First I enabled blindxfer in features.conf
Then in extensions.conf created an extension for conferences...it's 999
for me but it could be anything.
Then I added this NWayCall macro below. This is a modified version of
something I saw on Voip-info.org. When this macro is called, it first
checks to see if the caller was transfered to it or called the extension
directly. If they were transfered here, it gets the name of the SIP
user that transfered them, then checks to see if a conference with that
name exists. If the conference doesn't exist it creates one, otherwise
it adds the transferred person to the conference. If you weren't
transfered to this extension (as in, you called it directly) it adds you
to the conference.
Last time I tried this was last week, and I've been busy with other
things since. When I tried it, it worked but it was very twitchy. Any
improvements you can come up with would be appreciated.
Or if anyone has an entirely better way to do this, I'm listening.
exten = 999,1,Macro(NWayCall)
[macro-NWayCall]
exten = s,1,Noop(${BLINDTRANSFER})
exten = s,2,Gotoif($[${BLINDTRANSFER} !=
]?s-TRANSFERED|1:s-NOTTRANSFERED|1)
exten = s-TRANSFERED,1,GoTo(s-SIPHOLDER|1)
exten = s-SIPHOLDER,1,Cut(CONFHOLDER=BLINDTRANSFER,/,2)
exten = s-SIPHOLDER,2,Cut(CONFHOLDER=CONFHOLDER,-,1)
exten = s-SIPHOLDER,3,Goto(s-USERJOIN|1)
exten = s-USERJOIN,1,MeetMe(${CONFHOLDER},dwxM)
exten = s-USERJOIN,2,Hangup()
exten = s-NOTTRANSFERED,1,GoTO(s-SIP2HOLDER|1)
exten = s-SIP2HOLDER,1,Cut(CONFHOLDER=CHANNEL,/,2)
exten = s-SIP2HOLDER,2,Cut(CONFHOLDER=CONFHOLDER,-,1)
exten = s-SIP2HOLDER,3,Goto(s-CHECKCONFEXIST|1)
exten = s-CHECKCONFEXIST,1,MeetmeCount(${CONFHOLDER},CONFCOUNT)
exten = s-CHECKCONFEXIST,2,GotoIf($[${CONFCOUNT} =
]?s-INVALID|1:s-CONFNOTEMPTY|1)
exten = s-CONFNOTEMPTY,1,Gotoif($[${CONFCOUNT}
0]?s-HOLDERJOIN|1:s-INVALID|1)
exten = s-HOLDERJOIN,1,Meetme(${CONFHOLDER},qdAx)
exten = s-INVALID,1,Playtones(info)
exten = s-INVALID,2,Wait(10)
exten = s-INVALID,3,Hangup()
Hi all
How to configure adhoc conferencing in asterisk for
sip phones.pls give me if any document for that.
regards
ramakrishnan.n
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Re: [Asterisk-Users] Sipura 2000
If you unplug the ethernet cable on a Sipura SPA and then reset the power it'll boot up in a diagnostic mode. When you pick up the phone that's connected to it you'll get a dialtone and there are speical codes you can dial to do various things. Reset it to factory defaults by dialing followed by 73738# full instructions are here: http://www.sipura.com/Documents/faq/Section_3.html#4 Once you do that the provisioning enable should be no and you can reconfigure the device however it needs to be. Hi, Thanks for your response. I checked the setting, and indeed it was set to yes. However, once I change it to no and click on apply but after rebooting it's enabled again (with all settings reverted to factory defaults, as usual). Maxi. 2005/11/8, Rusty Dekema [EMAIL PROTECTED]: It's possible that your SPA-2000 is set up to read a configuration file from a remote host every time it boots up, which would overwrite any changes you make. If you log in as admin and go to the advanced view, there is an option under the Provisioning tab called Provision Enable. Make sure that this is set to no and your changes should remain in place. -Rusty On 11/8/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: Hello, I have a problem with my Sipura 2000. The problem is that it does not accept any change in the configuration. When I access to it, via browser or phone, and make any change, after clicking submit all changes all the changes I made dissapear and teh configuration remains with the original parameters. So I need to know how can I work it out. Thank you very much. Maxi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
