[Asterisk-Users] Where is the archive?
I've been trying to search the archives for older messages, but the archive at: http://www.mail-archive.com/[EMAIL PROTECTED]/maillist.html only seems to go back a few days. Is there another archive somewhere that goes back farther? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 800 Numbers (was Re: NuFone?)
I have been having the same problem with 800 numbers. NuFone and VoicePulse always behave the same (when one can't connect, neither can the other). I have so far found no explanation for this. Some other 800 and 877 numbers I can call. Can you elaborate on this at all? Thanks! o I couldn't dial 800 numbers via Nufone (IAXTel and PSTN worked) - I had forwarded him all pertinent information from my configs - All I got from support was, Everyone else can. and I can't reproduce it and we treat 800 the same as all other US calls, even after I had suggested that it wasn't him, it was the carrier he passes the call to. - I finally figured out that his carrier requires exactly 10 digits in the callerID, for tollfree numbers. This requirement does not exist for any other US number. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: 800 Numbers (was Re: NuFone?)
Ah, I was hoping to find the silver bullet, but no such luck so far. I have tried every combination of SetCallerID and SetCIDNum in my extensions.conf, both with and without the |a option, on both services with no luck still. When I call myself on our 877 number, I can see that the caller ID being sent is correct, 10 digits. Still won't call American Airlines. :( Well, there ya go. :)=20 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of=20 Robert Hajime Lanning Sent: Wednesday, March 17, 2004 1:15 PM To: Asterisk Users Subject: Re: [Asterisk-Users] 800 Numbers (was Re: NuFone?) =20 I did not have intermitent access. I could not dial any=20 tollfree number at all. It had to do with the CallerID I was sending. It needs to be=20 10 digits exactly. =20 quote who=3DMatt Lawson I have been having the same problem with 800 numbers. NuFone and=20 VoicePulse always behave the same (when one can't connect,=20 neither can=20 the other). I have so far found no explanation for this. Some other=20 800 and 877=20 numbers I can call. Can you elaborate on this at all? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I use the # key normally?
Is there a way to disable the transfer function of the # key? When calling other services, we often need it to access other menus, other voicemail, etc. Does this have anything to do with the T and t options in the Dial string? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail has hard-coded limit of 100 messages?
I got bit by this today and was surprised to see the limit of a measly 100 messages hardcoded into voicemail. Is that right or am I missing something? Obviously, this should be moved to voicemail.conf. Does anyone know if there's a reason why this hasn't been done, or if there's already a bug ID for it (I looked but didn't find one). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail has hard-coded limit of 100 messages?
Ah, so in a normal Asterisk world, the messages are supposed to be moved to another dir.? In our deviant Asterisk world, the voicemails are never checked through the phone, only through a custom web interface, so they stay in INBOX until they're deleted. Thus they collect quickly to over 100 total. Perhaps what you really should be asking is, why am I receiving more than 100 messages between the times when I check my voicemail, and what can I do to either decrease that number or arrange for voicemail to be checked more often? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10 msgs
I think international number dialed through voicepulse have to start with 011... (even if you're located in another countery). I asked them about that once, and that's what works for me (We've been dialing Spain and Germany recently, but never Japan) HTH, Matt --__--__-- Message: 4 Date: Sat, 21 Feb 2004 09:04:43 -0300 From: Daniel Bichara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse Connection Reply-To: [EMAIL PROTECTED] Hi, I have two * connecteds and I wish a phone connected to * #1 calls PSTN via Voicepulse connected to * #2, as follows: telephone --- Asterisk #1 Asterisk #2 Voicepulse When I dial 81-90... (japan), * #1 will route call to Voicepulse at *#2. Everything works between #1 and #2 but when #2 calls Voicepulse I get an error message: -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call rejected by 66.234.228.132: No such context/extension I am clueless!!! What could it be? Follow my confs... Exten.conf - *#1 exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) # Exten.conf - *#2 [outvoicepulse] exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _.,2,Congestion # Iax.conf - *#2 [voicepulse] context=VPWS secret=password auth=md5 type=friend host=66.234.228.132 disallow=all allow=speex allow=gsm jitterbuffer=no Daniel --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicepulse Connection
That's what I'm trying to get at. *normally* you expect to dial 00 but
when you're using voicepulse, Asterisk needs to start all international
number with 011. Think of it this way, in VoicePulse's mind, you're
always dialing from the US. Of course the user will try dialing 00
because that's the 'normal' way to do it.
So what you have to do is change your dial plan to intercept a 00
prefix and reformat it using a 011 prefix, something like this:
exten = _00[1-9].,2,Dial,IAX2/[EMAIL PROTECTED]/011${EXTEN:2}
Message: 9
Date: Mon, 23 Feb 2004 07:49:07 -0700
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10
msgs
Reply-To: [EMAIL PROTECTED]
In the UK it's 00 then the country code. So a call from the UK to my
phone would be 0013036742575.
Miie
Matt Lawson wrote:
I think international number dialed through voicepulse have to start
with 011... (even if you're located in another countery). I asked them
about that once, and that's what works for me (We've been dialing Spain
and Germany recently, but never Japan)
HTH,
Matt
-- __--__--
Message: 4
Date: Sat, 21 Feb 2004 09:04:43 -0300
From: Daniel Bichara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicepulse Connection
Reply-To: [EMAIL PROTECTED]
Hi,
I have two * connecteds and I wish a phone connected to * #1 calls
PSTN via Voicepulse connected to * #2, as follows:
telephone --- Asterisk #1 Asterisk #2 Voicepulse
When I dial 81-90... (japan), * #1 will route call to Voicepulse at
*#2. Everything works between #1 and #2 but when #2 calls Voicepulse I
get an error message:
-- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call
rejected by 66.234.228.132: No such context/extension
I am clueless!!! What could it be? Follow my confs...
Exten.conf - *#1
exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
# Exten.conf - *#2
[outvoicepulse]
exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = _.,2,Congestion
# Iax.conf - *#2
[voicepulse]
context=VPWS
secret=password
auth=md5
type=friend
host=66.234.228.132
disallow=all
allow=speex
allow=gsm
jitterbuffer=no
Daniel
-- __--__--
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[Asterisk-Users] Zombies got me!
After 2 weeks on bug #981 (Dropped Channels during dual redirect), I just posted 2 patches and almost have it fixed. The only problem is, the patch has a side effect of leaving some zombies. That is, the zombie channel that is created during the masquerade process doesn't get hungup. If I can get rid of the zombies, it will be 100% done. Out of 30 retests, I was left with 11 zombies, so it's not all the time. I suspect it's 66% of the time when a certain channel/extrachannel combination is used (11/15 is pretty close). Interestingly, that proportion seems to be the complement of the original 33% dropped channels. I would really appreciate help from someone who can help with this. Thanks. - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Zombies got me! - Fixed!
Never mind about the zombies. I fixed 'em real good... I didn't think I would find the solution so quickly. Thanks anyway. - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What can cause a Red alarm?
I know having it unplugged from the line will cause this, but it's not. It's an X101P single port FXO card. Most of the time it works fine but occasionally wigs out. In this case zttool shows a red alarm. Other times I call into it and it answers but I just hear a buzzing sound. In a day or two I'll try it again and it'll be back to normal I have an el cheapo POTS phone right there to make sure the line is good. There is a second, identical FXO card in the machine as well, which we haven't used in a long time. Thoughts? (Besides swap the two cards and try the other one) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please Explain newchan-pvt-pvt
I'm into my 4th or 5th day of working on bug #981. I know that part of the problem is that the fixup routine is called in chan_sip.c. Well in there is a line that says p=newchan-pvt-pvt. Problem is, that doesn't exist in this case. I see pvt described as private lock but that doesn't mean I have any clue what the ramifications are in this instance. Of course I can easily put a check into fixup that says if p is NULL then return success. But what does it *mean* if newchan-pvt-pvt is NULL? What should be done in this case? This situation happens (sometimes) when the dual redirect is used and it's in the process of transferring the original receipient. i.e. A calls B, then you do the dual redirect. Regardless of which parameter goes into Channel or ExtraChannel, B is the one that will cause the crash, as it's going through the masquerade process. Yes it is updated to cvs. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: speex with VoicePulse
Ours are setup to allow GSM or Speex, and I see that using VoicePulse it chooses GSM. Don't know the official policy. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] port number keeps changing
We have an asterisk installation that's on a residential-grade DSL and its port number (as visible from the outside) keeps changing, every time it registers. fuser indicates that asterisk is only using port 4569 for IAX2 (as it should), but when it goes out over the Internet, the port number is reported as something in the 1's and it changes every time. Obviously, this is a network issue of some type, but can anyone explain more precisely why this happens and how to stop it? We have some other installations that stay put on port 4569 like they're supposed to. Network issues aren't my area of expertise. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing 800 numbers with VOIP
Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800 numbers. Are 800 numbers treated differently somehow? Or is there a business reason for disallowing them? It makes the ringing sound but never connects. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe Video option
That's one of the things that's been on our (1control, I have nothing to do with Digium) wishlist/to do list that just hasn't gotten done yet. Currently, video in meetme is not supported. What we experience is the audio will conference with the other audio streams but the video just freezes. I was hoping to look into someday but I'm swamped with 1000 other things of higher priority. I have been thinking though, of some ways it could be supported, starting with the simplest and easiest: 1. First, if only 2 of the phones in the conference are video phones, allow them to exchange their video with each other, while having all of the audio streams conferenced as usual. 2a. The next step could be having each videophone rotate which stream it was showing for a few seconds (20 seconds maybe?). i.e. you could have 3 video calls mixed with several audio-only calls. Initially video call #1 would show #2's image, #2 would show #3's image, #3 would show #1's image for a few seconds, then rotate them by 1. Of course you don't need to show your own! :) Actually, ours has a picture-in-picutre in the corner so you can see yourself all the time anyway. 2b. The other option instead of time-rotating the images would be to try to show the image of whoever was talking. That kind of sounds like a pain to me, but maybe it's doable. 3. The really fancy thing would be to have Asterisk decode all of the video frames and create a 2x2 or 2x3 or 3x3 etc. mosaic, re-encode them and send them to each client. That REALLY sounds like a pain to me, but again, maybe it's doable. Right now I'd be pretty happy with 2a though. - Matt Message: 3 From: Regovich, Timothy [EMAIL PROTECTED] To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Date: Fri, 30 Jan 2004 13:07:46 -0500 Subject: [Asterisk-Users] MeetMe Video option Reply-To: [EMAIL PROTECTED] Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the v flag on my extension for the meetme app? Thanks, Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe Video option
No, there is no video output once the call goes into a meetme room. What I was talking about is a case where you have a regular video call between 2 video phones, then you try to send them to a conference room. The audio still works but the softphone's (Linphone in our case) behavior is to just freeze the video with the last image it received. I should mention one caveat to my previous suggestion (about just passing through the video with 2 phones or rotating through the images) ; I was assuming that all video calls were using the same format. In our case that would be true for the time being. That should be easy to do, just direct the rtp packets to the desired client. If they were using different video formats, you'd have to translate between them. From: Regovich, Timothy [EMAIL PROTECTED] To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: MeetMe Video option Date: Fri, 30 Jan 2004 16:02:55 -0500 Reply-To: [EMAIL PROTECTED] So you are actually getting the video to come out though? I am not getting any outbound video RTP traffic at all. What settings do you have? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: failover (was Re: voicepulse)
But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. Now there's an idea. I'm playing with this now, but there's at least 1 case I'm having trouble recognizing: The call connects but then drops due to unauthorized. It then only goes to the h extension and I don't get a chance to try again. Is there anyway to detect this? I have to cover all of the following cases: 1. VOIP IP address is not reachable. Goes to extension n+101 (seems to work as expected) 2. VOIP service answers but refuses with call with unauthorized. It just goes to the h extension Is there any watch to catch this failure? Perhaps put a timer on it and say if the call was less than 5 seconds or something try the next one? Yes I am using a correct username and password and getting this today (not from Voicepulse, from another provider). But there's also a moderate chance that during our systems' setup a name or password could be misspelled so I need to cover this case. 3. VOIP service connects but reports all busy. Well this one is hard to test. But I can make the Zap channel busy. It goes to extension n+101 as expected, so I'll have to assume that a busy VOIP service does the same thing. I was trying to determine if the t or h extension would be useful for these but I think not. The timeout has to be set long enough for someone to actually answer (20-60 sec or whatever). The h is always visited at the end of the call, whether it was sucessful or not. Any other cases, or suggestions how to handle case #2? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: failover (was Re: voicepulse)
OK, so I answered my own question. Turns out case #2 just goes to extension 2. Still trying to figure out the optimum arrangement so I don't have an inordinate number of extensions. Maybe like this: 1. First outgoing try 2. Second outgoing try 3. Third ougoing try 4. Play a message and/or hangup 102. Goto 2 203. Goto 3 304. Goto 4 But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. Now there's an idea. I'm playing with this now, but there's at least 1 case I'm having trouble recognizing: The call connects but then drops due to unauthorized. It then only goes to the h extension and I don't get a chance to try again. Is there anyway to detect this? I have to cover all of the following cases: 1. VOIP IP address is not reachable. Goes to extension n+101 (seems to work as expected) 2. VOIP service answers but refuses with call with unauthorized. It just goes to the h extension Is there any watch to catch this failure? Perhaps put a timer on it and say if the call was less than 5 seconds or something try the next one? Yes I am using a correct username and password and getting this today (not from Voicepulse, from another provider). But there's also a moderate chance that during our systems' setup a name or password could be misspelled so I need to cover this case. 3. VOIP service connects but reports all busy. Well this one is hard to test. But I can make the Zap channel busy. It goes to extension n+101 as expected, so I'll have to assume that a busy VOIP service does the same thing. I was trying to determine if the t or h extension would be useful for these but I think not. The timeout has to be set long enough for someone to actually answer (20-60 sec or whatever). The h is always visited at the end of the call, whether it was sucessful or not. Any other cases, or suggestions how to handle case #2? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicepulse
I was just about to write the same thing. It says busy. Is is REALLY busy or is something else wrong? This on the heels of switch-1.nufone.net being missing out of DNS. We have customers that expect their VOIP to work. Is there anybody that's reliable? I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broken DNS makes Asterisk whacky!
Check this out. I recently closed a bug I had written, #495 ExtraChannel in transfer causes crash Now I've been able to reproduce it, and somewhat narrowed down the culprit. But before I write another bug report, I wanted to see if anyone else had experienced the following (or would like to try:) When DNS (or outside connection to the network, not sure which) is broken and you have register= lines in iax.conf, Asterisk gets whacky. First of course you'll the message Host (whatever) not found at line (whatever) in iax2.c at startup. It takes a long time for the lookup to timeout. Later, I get some other generally bizzare behavior including: 1. I get everyone is busy at this time from devices that aren't. 2. The dual-redirect crash is 100% repeatable now. One of the last things you see is planning to masqerade 0sd8(*( INTO just before it crashes. If you remove the register lines from iax.conf (or fix your DNS/Internet problem) it runs fine. My first thought is it could be a timing issue due to the timeout taking so long during startup. Looking at the IAX2 registration code, it doesn't immediately appear that a failed registration would wreak any particular havoc. Does this sound similar to anyone else's experience? Anyone else care to verfiy? Our Asterisk version is pretty close to CVS, maybe a few weeks out but I didn't see any bugs listed that seemed to address this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 486 Busy message - SNOM 200
I have observed this also. Try downgrading the firmware on the Snom to 1.16x. That usually fixes it for me. Although that's obviously a workaround and not a true fix. It is the Snom phone sending the 486 message; I just don't know why. - Matt Hello All,=0A=0AWhenever I try calling SNOM 200, I am getting Everyone is = Busy at this time message though line is free. I tried with firmware image= of=0A2.02t and 2.03b (this I received from SNOM customer support) but prob= lem still exists. Rebooting Phone doesn't help either.=0A=0ASo, is it a fir= mware problem or there might be problem with the my asterisk configuration.= =0A=0AThanks for your time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Excessive VNAK's and jitter over IAX2
Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone --(ulaw over LAN)-- *1 IAX2 (ulaw over Internet) -*2(GSM over Internet) ---*3(ulaw over LAN)-- SIP phone Now what is shown below is the Asterisk in the middle, that is doing the conversion between the other two, one of which only speaks ulaw and the other only speaks GSM. The call basically seemed to work, except the audio quality was terrible, but it did seem to be basically connected. Asterisk started spewing out these VNAK messages, thousands of them as fast as it could. In the middle of it I did an IAX2 show channels to show what was in progress. The asterisk version shown here is a completely stock, CVS version from just a few days ago. The outboard Asterisks are somewhat modified but also re-synchronized with CVS within the last week. Also, all Asterisks have iax jitterbuffer=no. So, my questions are: 1. What do the excessive VNAKs indicate? Some type of communication error? NAT-related perhaps? 2. Does the 20,000+ jitter have something to do with the audio sounding terrible? 3. Why is there jitter at all if all Asterisks have their IAX2 jitter buffers turned off? 4. Is there any significance to the Username (none) for one of the peers? The Asterisk has both peer and user names for both machines. The caller name shows up, but the callee name is always (None) Ideas anyone? Thanks. DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK s Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter Format 24.9.xx.xxx i58 9/3 00015/6 0ms 0169ms ULAW 66.167.xx.xxx(None) 00010/4 8/00013 9ms 20743ms GSM 2 active IAX channel(s) *CLI DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Sphinx (Karl Putland)
On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: I wondered how long before someone started asking about that. That was all about creating a hook into Asterisk where it would stream the audio over a socket to another program (which was a custom program we wrote, but similar to the Sphinx example programs). The Sphinx program could do the voice recognition and possibly return a result to Asterisk over the socket, in the form of a text token. The goal was to be able to route calls using voice recognition. It basically worked, except at the time we were having terrible problems with the Zap cards and echo. It was distorting the sound so horribly that it wasn't even human-recognizable, much less machine-recognizable. We've put that project on indefinte hold for now but it could possibly be usable again in the future. It's been many months since I worked on it so I'm a little rusty. If you used something like an ISDN line where you could get a clear signal it did pretty well. Of course it was only choosing between about 10 different words. The eagi-sphinx-test program is basically a stub that has the necessary hook into Asterisk. We have a modified Sphinx AGI which is similar to the test program but just includes some of the logic of what we wanted it to do when it heard a token, etc.. As far as the Sphinx side of it goes, I actually posted the modification I made to Sphinx to read audio from a socket to the Sphinx mailing list. I was surprised that no one hardly seemed interested in it. The stock Sphinx server doesn't read audio over the network. It only communicates the responses over the network. That's why I had to make the above-mentioned audio-socket-read modification. - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help: codecs and bandwidth
In an attempt to reduce bandwidth usage, I tried forcing my Asterisk to use Speex. I did a disallow=all then an allow=speex. The crazy thing is, it didn't reduce the bandwidth usage at all! I can do an IAX2 show channels and it shows the call being in format 512 (Speex, right)? Then I switch iax.conf to only allow ulaw. I retry the call. IAX2 shows the call is in format 4, as expected. But in both cases, iptraf shows the call is consuming about 170kbps (85 each way) over the baseline. How can this be? How can there be no difference between the two? :-/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help with jitter buffer/quality settings
I'm using Asterisk to do audio as well as H.263 video over SIP. Actually the video works pretty well but I have trouble with the audio. I'm wondering if someone can suggest codec/jitter settings or other tweaks. The system looks like this: Softphone ---ulaw Asterisk #1 --IAX (usually GSM)- Asterisk #2 IAX (usually GSM) Asterisk #3 --ulaw- softphone. Now originally I had all jitter buffers turned off. Today I tried turning the jitter buffers for Asterisks #1 and #3 on with maxjitterbuffer and maxexcessbuffer set to 1000. That seemed to help a little. Then I tried those settings for Asterisk #2 and it seemed the same or worse. I have tried several different audio codec choices as well. The puzzling thing is, the video works pretty well but the audio has the trouble. It's largely bandwidth-related but a residential DSL line ought to be able to get out a good audio signal! Sometimes the audio is good but most of the time it's garbled or cuts out completely. We've tried reducing the bandwidth of the video signal, but that's not the whole story. Now when I do an IAX2 show channels on Asterisk #1 or #2 (during a call), I see Lag and Jitter values in the tens of thousands and climbing, in spite of the fact that I set jitterbuffermax to 1000. That also is puzzling. I suspect I shouldn't be seeing large numbers like that. We also do audio-only calls of course. Sometimes those sound great but a lot of times those break up as well. So, my Q's are: 1. What should I set the jitter buffers settings for each of the 3 Asterisks above? The links between them may be low or high speed. 2. Should I force a particular audio codec? 3. Is there anything else I can do to improve the audio quality? (besides a faster connection) Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo cancel in MeetMe?
I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into white noise. Which part of the software is responsible for echo cancellation in a MeetMe room? Is it a setting on the phones themselves, or within Asterisk? And is this related to echo cancellation on the POTS lines? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancel in MeetMe?
Oops, my bad. Turns out it was just mixer settings, feeding back through the soundcard. Sorry for the noise. Message: 14 Date: Wed, 03 Dec 2003 17:43:16 -0500 From: Matt Lawson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo cancel in MeetMe? Reply-To: [EMAIL PROTECTED] I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into white noise. Which part of the software is responsible for echo cancellation in a MeetMe room? Is it a setting on the phones themselves, or within Asterisk? And is this related to echo cancellation on the POTS lines? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX port numbers?
I see that when an Asterisk connects to another one via IAX, it seems to use port 4569 for the first one. But if it has multiple IAX connections the additional ports seem to be chosen at random. Is there anyway to predict, or specify which ports or range of ports to use, for the sake of setting up a firewall? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Solved! Snom 200 Busy signal
Now that you mention it, I did observe the PUBLISH message. Can someone please tell me exactly the change that was made that fixed this? (file/lines) I can do a diff -r and see a few changes from CVS but I'd like to be sure. We have a lot of custom changes as well so it's non-trivial to update to CVS. I would definitely like to check on this one patch though. Thanks. - Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson Sent: Thursday, November 20, 2003 7:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Solved! Snom 200 Busy signal As a follow up to my earlier posting, the problem with the Snom 200 Busy signal was the firmware! I reverted back to 1.16x and everything's OK - Matt I had the same type of problem you did when I first upgraded to 2.02t. But I did a CVS update on 11/19/03, and everything is fine now. What I did notice (and I have a customer that saw the same thing) is that that been able to determine what they're publishing with it. But Asterisk didn't seem to like it until I did a CVS update. It didn't seem to respond to it. Now Asterisk sends back a Status: 405 Method Not Allowed message, and everything is fine. I can't definitively say this was the cause because I didn't go back to the old Asterisk code and verify. But I can tell you that my Snom 200 phones with version 2.02t work fine now. Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 200 stuck on Busy
With a recent update to Asterisk CVS, and versions 2.02r and t of the Snom 200 firmware, I'm getting the Snom phones stuck reporting Busy.: -- Got SIP response 486 Busy Here back from 10.12.34.248 -- SIP/3064-b07d is busy They're on-hook, not doing anything. They are registered fine. I can pick them up and call out over Zap or IAX fine. I just can't call the extensions because they always report busy. Has anyone else observed this? - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Solved! Snom 200 Busy signal
As a follow up to my earlier posting, the problem with the Snom 200 Busy signal was the firmware! I reverted back to 1.16x and everything's OK. That made today pretty complicated, since I already had a new kernel and a new Asterisk build I was trying all at once Hopefully someone else will benefit... - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO card still won't pick up...
I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial(SIP/3064-564c, Zap/g1/ww954...) in new stack NOTICE[245776]: File app_dial.c, Line 698 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time When I try to call in, usually nothing happens. One time it answered and Asterisk acted like it was going through the dial plan steps, but on the phone I never heard anything. It just hung up immediately. After a couple more seconds Asterisk noticed the hangup and stopped its dial plan. chan_zap.so loads OK, and I can do this: *CLI zap show channels Chan. Num. Extension ContextLanguage MusicOnH 1incoming31 en 2incoming31 en Here's an exceprt from when Asterisk starts: [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Not found (No such file or directory) [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 1, with 0 conference users -- Registered channel 1, FXS Kewlstart signalling DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 2, with 0 conference users -- Registered channel 2, FXS Kewlstart signalling == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Registered application 'CallingPres' -- Registered to '172.16.0.1', who sees us as 172.16.255.157:4569 WARNING[147466]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable [pbx_config.so] = (Text Extension Configuration) ZTCFG seems OK: # ./ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. Zaptel.conf is blank. Zapata.conf follows. These were the same way when the older version was working: # cat zapata.conf [channels] echocancelwhenbridged=yes echocancel=yes stripmsd=1 callerid=asreceived language=en context=incoming3121 signalling=fxs_ks rxgain=3.0 txgain=0.0 usecallerid=yes group=1 channel=1 echocancelwhenbridged=yes echocancel=yes stripmsd=1 usecallerid=no callwaiting=no callerid=intercom 9876543210 context=incoming3130 language=en signalling=fxs_ks group=1 channel=2 So, I'm wondering. 1. What other diagnostic steps can I do to narrow this down? 2. What flags/switches in Asterisk or Zaptel could be causing this, particularly if they've changed recently? 3. Is there a 'how to diagnose zap problems' guide anywhere? Thanks. - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_zap won't load after CVS update
Ah ha. That's *almost* got it. It will now load and * will run. The only big gotcha is it won't pick up or dial out on a POTS line. ztcfg shows both channels configured OK, as does 'zap show channels.' If I try to dial out I get: -- Executing Goto(SIP/3063-74d0, outside|9555|1) in new stack -- Goto (outside,9555,1) -- Executing Dial(SIP/3063-74d0, Zap/g1/ww954...) in new stack == Everyone is busy at this time I had the same problem the other day. Resolved it by essentially blowing away the existing src directories (rename them if you want) and doing a new cvs checkout. I know I seen someone post something a month or two ago relative to using a cvs flag to effectively over-write everything on the local system. Guess we're all supposed to be linux cvs experts. In my case, I was simply trying to do an update to asterisk (without zaptel, zapata, etc), and that didn't work since some constants were apparently defined in some non-asterisk header file (zaptel, I think, but I didn't bother to search it out) that were needed in one of the channels modules. Sounds like you've bumped against the same issue. Regards Mick West --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_zap won't load after CVS update
I've just finished updating my Asterisk to the CVS version. Unfortunately, chan_zap won't load anymore. The hardware has not changed and the config files have not changed. I can re-install the two packages back and forth. The old one will still work. The new one won't. I tried updating to a brand-new zaptel and wcfxo modules, with no difference. This has got to be the most frustrating thing about dealing with Asterisk. This is also the same error I get trying to get the FXS cards to work (I have never succeeded). There must be something else in the Makefile or configuration files. Is there anything different regarding zap interfaces in the config files since maybe 3 months ago? The other differences I noticed were the modules chan_alsa.so, chan_oss.so (which didn't appear to be there before, or maybe in a different order), and a new requirement for libpri.so Same error message: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 1, with 0 conference users ERROR[16384]: File chan_zap.c, Line 5287 (mkintf): Unable to get span status: Inappropriate ioctl for device ERROR[16384]: File chan_zap.c, Line 6838 (load_module): Unable to register channel '1' WARNING[16384]: File loader.c, Line 305 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 400 (load_modules): Loading module chan_zap.so failed! zapata.conf: [channels] echocancelwhenbridged=yes echocancel=yes stripmsd=1 callerid=asreceived language=en context=incoming3121 signalling=fxs_ks rxgain=3.0 txgain=0.0 usecallerid=yes group=1 channel=1 echocancelwhenbridged=yes echocancel=yes stripmsd=1 usecallerid=no callwaiting=no callerid=intercom 9876543210 context=incoming3130 language=en signalling=fxs_ks group=1 channel=2 zaptel.conf can be blank or: loadzone=us defaultzone=us fxoks=1-2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with include files current CVS
Hello, I'm trying to compile a brand new CVS Asterisk and running into trouble with include files. I have an older version of Asterisk that I can compile (2-3 months old) that I can compile fine, but the new one gives me this: make[1]: Leaving directory `/home/matt/asterisk_update/stdtime' if [ -d CVS ] ! [ -f .version ]; then echo CVS-10/01/03-13:06:31 .version; fi gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -I./include/asterisk -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-10/01/03-13:06:31\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c asterisk.c: In function `set_priority': asterisk.c:377: storage size of `sched' isn't known asterisk.c:384: warning: implicit declaration of function `sched_setscheduler' asterisk.c:384: `SCHED_RR' undeclared (first use in this function) asterisk.c:384: (Each undeclared identifier is reported only once asterisk.c:384: for each function it appears in.) asterisk.c:392: `SCHED_OTHER' undeclared (first use in this function) asterisk.c:377: warning: unused variable `sched' make: *** [asterisk.o] Error 1 It seems that the structure sched_param is not defined anywhere. I did find a definition of sched_param in /usr/include/linux/sched.h but that file is not included. When I tried to include it, I ended up with one include file after another I was having to add, and then they were conflicting over redefinitions, etc... Where is Asterisk normally expecting to find the definition of struct sched_param? Thanks - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transferring to Meetme
Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can subsequently transfer it to the meetme? All of this must be done through the management interface and not require users pushing any button.s Thanks. - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No 'ringing' sound to outside callers
Most of the time, when someone calls in from the outside on a POTS line, and possibly over IAX as well, they don't hear any ringing sound while the internal SIP phones ring. If you call from an inside SIP phone, even forcing it into the incoming context, you hear the ringing. The outside calls can answer and talk fine; just no ring indication. Is there a setting that controls this? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Turning a regular call into a conference?
What steps would have to happen, in order to take an already-connected call and move both parties into a conference room? i.e. do both parties have to be parked first, or can one or both of them just be immediately transferred to a MeetMe extension? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Loop counter variable in dialplan?
How can I loop through something x number of times in the dialplan? i.e. if I get an invalid extension I want to re-play the menu, but not forever. Maybe 3 tries or something. I'm pretty sure that I've seen it before, where you can increment a variable and do Gotos based on it. But I've searched the Asterisk handbook, searched the user archives, and Googled for it, and can't find it now. Anyone have a link with an example of this? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alternatives to FXS cards?
Hi everyone, I know someone makes a product that's a POTS phone to SIP converter, where you just plug your POTS phone in one side and the network cable in the other. Has anyone successfully used any of these with Asterisk, and if so how expensive were they? I ask partly out of frustration with the FXS cards but mostly because it would make installation MUCH easier for what we're doing, plus it would be another piece of hardware that we could re-sell, plus it would free up some slots in the server, which is valuable real estate. Comments? TIA - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto-detect of fxo vs. fxs channels?
Is there a way to determine which channels belong to fxo vs. fxs devices? I need to write an auto-configuration program that can match up channel numbers to types. I have to assume there's an unknown ordering of fxo and fxs cards. Suggestions? TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * Picks up line during outgoing call
We have some regular POTS phones connected to our incoming line as well as the machine that runs Asterisk. Sometimes during an outgoing call from the POTS phone, the Asterisk will pick up also, and play its menu. The FXO card is set to fxs_ks signalling; I'm told this might be the culprit but I really don't understand about the signalling types and what the ramifications of different ones are. Any help? Thx. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Noisy/Clicky hangup
When I call in from an outside POTS line to a Zap channel, and the call ends, it seems like the hangups are very sloppy. I see Asterisk give the hangup command, but on my phone there's lots of clicks and the line acts like it's staying open for several seconds, then I hear a phone ringing sound followed by If you'd like to make a call, please hang up and try again... Is there something wrong with my setup that it acts this way, or is that just how it is? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_zap Cannot handle frames in 2 format
I have discovered something quirky in our Asterisk. If I call in to a Zap channel (from an outside POTS line), then transfer the call around several times, I get the above error, after which it will hangup. I believe Asterisk may issue a SIP CANCEL to the extension it was starting to dial. Now when I say 'transferred around several times,' our routing is pretty compex and uses the database lookup for user extensions. It plays a static message, then goes to a 'which user do you want' type menu, then may go to voicemail or ring an extension, while I beat on it with Redirect commands through the management interface. I sometimes redirect it to specific SIP extensions and sometimes to users, which have to be looked up in the database. The SIP phones are set to communicate with Asterisk only using mu-law. The IAX connection uses GSM (and we do have multiple Asterisks talking over IAX). One thing that puzzles me is that the format 2 in chan_zap I believe corresponds to GSM. Where is it getting that? It should only be using mu-law on the local system. The only other possibility that occurs to me, is that voicemails are left in GSM format. Is it possible that if a call gets transferred after it's already in the process of leaving a voicemail that will break it? Suggestions? Thanks, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
