Re: [Asterisk-Users] DTMF and ivr systems
On Thu, Jun 29, 2006 at 10:42:05AM -0700, Shane wrote: Ther's probably a simple answer to this but I've searched around and haven't located anything as yet. Is there a way to have DTMF tones passed through Asterisk without it messing with them? I am using a tdm21b card and when I call an ivr system from the telephone handset (routed over sip or iax2) such as telebanking, the ivr has trouble recognizing tones. When I tested this with a remote party, I was told tones were breaking up. For example, a long press would result in a click, some silence and a small dtmf on the remote end. Triggering a speed dial didn't go well either as he heard only a few tones. I have dtmfmode=inband in sip.conf and have tried relaxdtmf=yes in zapata.conf. I realize Asterisk does need to detect dtmf for things like call parking but can it just pass the audio to the other side with no regard for whether it's dtmf digits? IE. long press results in long tone, etc. We've run into some problems with * and dtmf. *Usually* dialing the digits more slowly will help. It seems to be worse with some SIP devices than others. But when calling from my handset on an FXS to my * system which goes IAX2 to another * system and then out a VoIP service line, I can dial as fast as my fingers can go and never miss... snoms and sipuras on the latter system can have trouble if we dial too quickly... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 hardphone for Asterisk?
On Mon, Jun 26, 2006 at 12:08:48AM -0400, Doug Crompton wrote: Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. 1. If you have *, you don't necessarily need multiple handsets off of one base. 2. Cordless phones also require power 3. If the multi-handset cordless phone does suit your needs best, then get a SIP ATA device like a Sipura or IAXy and you should have your needs met. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] SE Michigan asterisk users group
Our main office is near Lansing, but we have a person who lives in the AA are that would like to attend such a group. On Thu, Jun 22, 2006 at 04:27:02PM -0400, BerkHolz, Steven wrote: I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] latest @Home questions
We are moving our asterisk 1.0 system to a new Asterisk @Home system (2.8) and I am the one in charge of doing it. I have run into a snag, though, on meetme conferences and with the transfer key. Regarding the transfer, it appears that both directions of all calls can transfer by pressing the # key. I do not like that ability. I would like to change it by doing 2 things: 1. Make the transfer sequence be ## rather than # I looked at the features.conf file and it didn't have an entry for blindxfer, so I added it. However, # is still the transfer character so it doesn't seem to be recognizing the settings from features.conf. 2. Not allow incoming calls to transfer at all. I've looked at the dial() string on incoming calls and they do not contain a t or T like I would expect for the channel to be able to transfer. As for the meetme conferences, the docs say that for all extensions defined, there is a meetme conference at 8ext. So extension 250 would have a conference at 8250. This isn't the case on our installation. I went to the Conference menu item and defined a conference at 1000 and I put an entry into the IVR for incoming callers to get into the conference and that works fine (except that after the PIN number, they cannot press # to signal the end of the PIN -- that will try a transfer). However, I don't have a way to get to the conference from an extension. I think I'm missing something, because meetme setup cannot be that difficult... Thanks for any help anyone can offer. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] TDM4xxP
On Sat, May 06, 2006 at 12:45:10PM -0400, Sean Cook wrote: hm... why not just use ztdummy and save the $150 for the card? ztdummy can have timing inaccuracies that will be removed with a Digium card. Am dealing with that right now on our company system... Steve Totaro wrote: I have a TDM4xxp card with no modules. My question is, will this card be sufficient to provide timing or does it need to have modules? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] Fwd: meetme conference latency degrades...
I haven't seen this appear on the list, so I thought I would resend it... Sorry for the repost if it did appear before... - Forwarded message from Michael George george - Date: Wed, 3 May 2006 21:48:09 -0400 From: Michael George george Subject: meetme conference latency degrades... To: asterisk-users@lists.digium.com We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave the conference and return, the latency is unnoticable again. The load on the box is minimal, and only our meetme is running most of the time. Checking system load with top shows 0.1 or less. We have no digium hardware and use ztdummy for our timing device. zttest yields results generally in the area of 99.96%, but about 3-4% will be as low as 95%. In much smaller systems with Digium hardware, the accuracy is never below 99.98% and is often 100%. Is this apparent inaccuracy of the ztdummy timer likely the cause of the increasing latency in our meetme conference? Is there any way to improve it? Thank you, in advance, for any help. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] meetme conference latency degrades...
We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave the conference and return, the latency is unnoticable again. The load on the box is minimal, and only our meetme is running most of the time. Checking system load with top shows 0.1 or less. We have no digium hardware and use ztdummy for our timing device. zttest yields results generally in the area of 99.96%, but about 3-4% will be as low as 95%. In much smaller systems with Digium hardware, the accuracy is never below 99.98% and is often 100%. Is this apparent inaccuracy of the ztdummy timer likely the cause of the increasing latency in our meetme conference? Is there any way to improve it? Thank you, in advance, for any help. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] nic aliases not working
I have an * box that I need to chang the IP address on. My hope was that I could add an alias to the interface with a different IP address, have * bind to all addresses, change DNS and when no more hits come on the old address. However, IAX registrations coming in to the alias don't seem to get acknowledged by *. Even with iax2 debug on, I don't see any attempts. We can ssh in on both IP addresses and I have bindaddr=0.0.0.0 in iax.conf. Is this not possible for some reason? Maybe multiple IP addresses work but nic aliases do not? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] nic aliases not working
On Tue, Apr 11, 2006 at 02:08:09PM -0700, Daniel Hazelbaker wrote: Have you quit and relaunched Asterisk? (not a reload, but a full quit process and restart) I know in the past when I have a process already listening to 0.0.0.0 it will not always pick up a newly added NIC alias address without re-binding. Yes. I set bindaddr=0.0.0.0 and I have done a complete start and stop. I may try again just to be sure, though... Prior to my putting 0.0.0.0 into the file, there was no bindaddr setting. I am not sure if the default is to bind to eth0 or to bind to all... On Apr 11, 2006, at 12:21 PM, Michael George wrote: I have an * box that I need to chang the IP address on. My hope was that I could add an alias to the interface with a different IP address, have * bind to all addresses, change DNS and when no more hits come on the old address. However, IAX registrations coming in to the alias don't seem to get acknowledged by *. Even with iax2 debug on, I don't see any attempts. We can ssh in on both IP addresses and I have bindaddr=0.0.0.0 in iax.conf. Is this not possible for some reason? Maybe multiple IP addresses work but nic aliases do not? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] digium.com redesign
I may be way behind here, but I see that digium redesigned their site. I cannot find the mailing list search screen. I have found the mailman list page, but that doesn't have have a nice search ability. Do I need to just rely on google and other generic search engines or is there a search on the digium site? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 contexts being confused
On Wed, Mar 01, 2006 at 10:39:36PM +0100, Rene Kluwen wrote: I have the same problem. My solution is differentiate in extensions.conf, since all calls are terminated to different MSISDN's. So in extensions.conf I have something like: [incoming] exten = 9995551212,1,Goto(company1-context,s,1) exten = 9995551213,1,Goto(company2-context,s,1) I have done something similar. So the calls are being handled correctly as the progress through the system. The problem is that when doing a show channels all connections from either trunk are indicated as being from the last one in sip.conf. Also, and this is the bigger problem, I don't have the control over codecs that I would like. To allow 729 for one trunk and ulaw for the other requires that both trunks be defined to allow both codecs and hope that the VoIP provider will always prefer the right one. Question for you: In your sip.conf for your two SIP trunks, do they use any authentication (username and password) or do they use IP exclusively to determine the trunk? Thank you for your reply! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael George Sent: woensdag 1 maart 2006 15:48 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP contexts being confused I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls. The sip contexts involved are: [Cust1_in] canreinvite=no context=incoming fromdomain=voip.provider.net host=voip.provider.net fromuser=9995551212 username=9995551212 nat=no type=friend disallow=all allow=g729 musiconhold=Cust1 accountcode=Cust1 amaflags=documentation [Cust2_in] canreinvite=no context=incoming fromdomain=voip.provider.net host=voip.provider.net fromuser=9995551213 username=9995551213 nat=no type=friend disallow=all allow=ulaw musiconhold=Cust2 accountcode=Cust2 amaflags=documentation There is no SIP registration involved because the service provider knows the address of the PBX server and will contact that address for calls on either trunk. I'm not sure that fromuser and username are even being used by the provider or *. However, *all* calls coming in for either account are reported in show channels and the verbose output as being from the second context. Also, * is negotiating the codec based on that latter context, so when the channels should be 729, they are being negotiated as ulaw. I was tempted to blame the provider, but if I change the order of these two entries in sip.conf, then the Cust1_in context is used. All calls appear as coming from that SIP channel and the incoming calls fail because it will only allow 729 and the provider only allows ulaw. [I have analogous Cust1_out and Cust2_out contexts for outgoing calls, but they seem to work fine.] It's almost as though the call comes in from the provider and the IP address is looked up in a table to find the context that applies. Asterisk then looks for an entry with that IP address, finds the last of these two in sip.conf, and uses that for the incoming channel. So it appears that we cannot have two customers with this SIP provider and keep things straight. It is possible that the problem is a shortcoming with the provider, but it's looking more like a shortcoming with *. Can anyone help me by offering a solution and/or explanation of what's happening here? If I need to provide more information, I'd be happy to. I'm sure the vPBX concept will work well, but this problem is holding everything up. If I cannot fix it, we cannot continue selling slots on the vPBX. Thank you! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP contexts being confused
I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls. The sip contexts involved are: [Cust1_in] canreinvite=no context=incoming fromdomain=voip.provider.net host=voip.provider.net fromuser=9995551212 username=9995551212 nat=no type=friend disallow=all allow=g729 musiconhold=Cust1 accountcode=Cust1 amaflags=documentation [Cust2_in] canreinvite=no context=incoming fromdomain=voip.provider.net host=voip.provider.net fromuser=9995551213 username=9995551213 nat=no type=friend disallow=all allow=ulaw musiconhold=Cust2 accountcode=Cust2 amaflags=documentation There is no SIP registration involved because the service provider knows the address of the PBX server and will contact that address for calls on either trunk. I'm not sure that fromuser and username are even being used by the provider or *. However, *all* calls coming in for either account are reported in show channels and the verbose output as being from the second context. Also, * is negotiating the codec based on that latter context, so when the channels should be 729, they are being negotiated as ulaw. I was tempted to blame the provider, but if I change the order of these two entries in sip.conf, then the Cust1_in context is used. All calls appear as coming from that SIP channel and the incoming calls fail because it will only allow 729 and the provider only allows ulaw. [I have analogous Cust1_out and Cust2_out contexts for outgoing calls, but they seem to work fine.] It's almost as though the call comes in from the provider and the IP address is looked up in a table to find the context that applies. Asterisk then looks for an entry with that IP address, finds the last of these two in sip.conf, and uses that for the incoming channel. So it appears that we cannot have two customers with this SIP provider and keep things straight. It is possible that the problem is a shortcoming with the provider, but it's looking more like a shortcoming with *. Can anyone help me by offering a solution and/or explanation of what's happening here? If I need to provide more information, I'd be happy to. I'm sure the vPBX concept will work well, but this problem is holding everything up. If I cannot fix it, we cannot continue selling slots on the vPBX. Thank you! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 SIP registration situation
I have 2 Polycom SP 500's attached to my system. Both are behind NATs, but both seem to work fine, for the most part. A few weeks ago, I started to notice that I get an error message from one of them: Feb 20 08:54:58 NOTICE[10663]: chan_sip.c:7691 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for 'zzz.aaa.103.75' However, trying to call the phone works fine, so I know it's registered. Turning on SIP debug for the phone shows that it attempts to register and is rejected with unauthorized, then another attempt is rejected with forbidden, and finally a registration succeeds. The other phone doesn't exhibit this behavior. I am running asterisk 1.0.7. Has anyone used Polycoms remotely from behind a NAT enough to have insight as to what is going on? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP501 Endless Loop
On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED] wrote: I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003 FTP service, WSFTP server and a few other Windows based FTP servers. Anybody have an idea as to how to get around this? I cannot get support on this phone (Polycom tells me to call the reseller and the reseller won't touch it for less than $95/hour). I second the suggestion to use your * system as the FTP server. You can then get more information from the logs and you have more choices for which server to run. We have excellent luck with pure-ftpd in a gentoo installation. It allows us to make virtual FTP users so that *only* the users who are defined in the virtual system are able to open a session. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 netboot not working.
Are you sure you have the FTP server's IP address set correctly in the phone's configuration? On Thu, Jan 05, 2006 at 05:17:41PM -0500, Ken D'Ambrosio wrote: Anthony Rodgers wrote: Is the mac-address.cfg file name in lower case? Yeah, it is. Hell -- I've cut-and-pasted the filename from the below logfile, and been able to FTP it just fine. I've run an ethereal dump, and it never even -asks- the server for the file, so I'm kind of confused there. I've reset the phone with 4-6-8-* keys, but same thing. I'm tempted to try another phone, and see if I get anywhere. But before I -kill- another phone, I thought I'd ask if anyone else has seen this or anything like it... -Ken On Jan 5, 2006, at 1:37 PM, Ken D'Ambrosio wrote: When I try to boot my 501, it runs through the usual stuff, then stops with Config file error Error is 0x4020 and then reboots. The log on the FTP server shows: 0105164151|app1 |3|00|Bootline: ircaIP 0105164155|cfg |3|00|Image bootrom.ld has not changed. 0105164159|cfg |3|00|0004f202f803.cfg could not be downloaded, getting next file. 0105164206|cfg |3|00|Image sip.ld has not changed. 0105164237|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0105164237|app1 |6|00|Uploading boot log, time is THU JAN 05 16:42:38 2006 I can't figure out why it can't download the cfg file -- the permissions are right, etc. I can FTP all the files as PlcmSpIp (with PlcmSpIp as the password) just fine. It -does- try to d/l the .cfg file, but appears to ignore it, even when I give it extension-specific config info (gives the same error). Any ideas? I'm afraid to try to provision my other phones, for fear of winding up in the same spot. Thanks, -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] SNOM 360 locked up SOLVED
On Thu, Dec 22, 2005 at 03:58:07PM -0800, Steven Ringwald wrote: Thank you so much for your help, Christian! Your suggestion worked perfectly, and the phones came back up without a problem. What part of his suggestion? Upgrading the firmware to 4.5 via the tftp server? Please elaborate for the benefit of others who may run into this problem. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] Aastra firmware 1.3.x
I have a related question about the 480i and firmware 1.3... I have a 480i that I got about 1yr. ago and it didn't work well at all. I finally got around to updating the firmware. However, the phone will not load its firmware. I set up the tftp server and I pointed the phone to it. I can watch the logs on the tftp server and see that the transfer initiates. However, at a point in the startup process, the Aastra locks up. The little progress wheel on the display freezes and it won't respond to anything. Not the keypad, Web interface, not even (IIRC) pings. Has anyone run into this before? On Mon, Dec 26, 2005 at 01:02:13PM +0100, BennyBad wrote: Using the: # headset tx gain: # headset sidetone gain: handset tx gain: 10 handset sidetone gain: 0 # handsfree tx gain: 2 Worked great for Me ! Actually we have 10 480i's and the settings are not the same for all phones. handset tx gain xx varies form +5 to +10, to get the same result. So I believe this is a HW issue. Reg. BennyB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: 24. december 2005 04:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue) Taco Scargo wrote: Hello, Just bought two 480i's which I updated to firmware 1.3 I experience the 'Far-End sound level issue' now. I tried configuring the handset tx gain: value but can only make it sound softer, not louder. If there is someone that has managed to get decent Far-end sound level, could he or she please e-mail their used values ? I have a similar issue with the Aastra 9133i and recorded .wav voicemail files. The recorded wav is too soft. I need to find a way to boost the volume level. Does anyone have any solutions or ideas? ___ --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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] Aastra 480i
I have an Aastra 480i that used to have firmware 1.0.0... on it. I got the new 1.3 firmware and had the phone fetch that from my TFTP server, but after running about 15s, it stops. No more downloading, no response to WebGUI, no response to buttons, nothing. I rebooted it (not a good idea, I know) and it complained that there was no application and tried to reload from tftp server again. Same thing happened. So I tried the firmware for the 480i CT IP phone, but that did the same thing. The little wheel spins on the display when it boots but it stops when the download stops. Anyone have any advice? I just want the firmware in there, I am happy to manually configure the phone with the UI... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 behind the NAT
On Tue, Dec 13, 2005 at 11:32:15PM -0500, Tom Rymes wrote: On Dec 13, 2005, at 8:25 AM, Michael George wrote: I have a similar problem with a client's system. They have * 1.0.x behind a NAT with all the SIP phones on the local network. Their VoIP provider is outside the NAT (a Metaswitch at their ISP, connected to the phone lines from there). Their network guy has the firewall passing traffic on ports 5060 and 1-2 to the * system. I have externalIP and localnet set, but nat=no (default) is the case for this one. Occasionally they will place outgoing calls and the other party does not hear anything. Usually another attempt at the call will pass audio normally. One person who makes about 100 calls a day remembers having this happen on about 7 calls one day. No one recalls this ever happening on incoming calls (though this client primarily makes outgoing calls, I believe). Apparently this has been happening for a while and they just now mentioned it to me. Would nat=yes in the general section of sip.conf make a difference in this case? Is there anything else I could look at that might alleviate this problem? Without being a smartass, the only way to find out is to see if it works. More obviously, if the Asterisk server has a NAT between it and the ITSP, then use nat=yes, if it doesn't, then use nat=no. Of course, if you set nat=no, then don't bother setting localnet or externip, either. Also keep in mind that some routers' DMZ settings still leave your box behind NAT. They just forward all of the ports to the specified address. (Linksys routers do this.) I didn't detect any smartassity in your response... I'm going to try nat=yes in the general section and then I'm going to trim down the RTP port range just for fun and see what happens. Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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 behind the NAT
On Fri, Dec 09, 2005 at 03:23:31PM -0500, Tom Rymes wrote: On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi: i added these two lines to my general context ,but nothing happened the same result the sound came in one way for 3 seconds and stopped but it didnt hangup. --- Jeffery Chen [EMAIL PROTECTED] wrote: If your Astersik server behind NAT too, your need modify SIP.conf like this externalIP= x.x.x.x localnet= x.x.x. hope this can help you Make sure that you have ports 5060 and ports 1-2 UDP forwarded to your Asterisk server. (Asterisk uses UDP for SIP, not TCP!!!) Also, in addition to the externip and localnet entries in sip.conf, You need to add a nat=yes entry I have a similar problem with a client's system. They have * 1.0.x behind a NAT with all the SIP phones on the local network. Their VoIP provider is outside the NAT (a Metaswitch at their ISP, connected to the phone lines from there). Their network guy has the firewall passing traffic on ports 5060 and 1-2 to the * system. I have externalIP and localnet set, but nat=no (default) is the case for this one. Occasionally they will place outgoing calls and the other party does not hear anything. Usually another attempt at the call will pass audio normally. One person who makes about 100 calls a day remembers having this happen on about 7 calls one day. No one recalls this ever happening on incoming calls (though this client primarily makes outgoing calls, I believe). Apparently this has been happening for a while and they just now mentioned it to me. Would nat=yes in the general section of sip.conf make a difference in this case? Is there anything else I could look at that might alleviate this problem? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 error: Ouch ... error while writing audio data: : Broken pipe
On Fri, Nov 18, 2005 at 10:22:23AM -0600, Kevin P. Fleming wrote: Leo Burd wrote: Any ideas about what is going on? Yes. You didn't read the warnings prominently displayed at the end of 'make install' about removing old modules from /usr/lib/asterisk/modules. Does that include the 729 codec modules, or can they stay there for 1.2? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
Re: [Asterisk-Users] Music on hold not initiating RTP stream?
On Fri, Sep 30, 2005 at 06:47:51PM -0500, Kevin P. Fleming wrote: Ray Van Dolson wrote: The ATA's are Sipura SPA-2002's and I have MOH Server set to 899 on each. Take that out, you don't need it. He had this in there for testing to show that the problem was not mpg123, which he did. However, with a call in progress, if I hit hold or flash on SIP ATA 1, it behaves correctly, but no music on hold is heard on SIP ATA 2. I can see in my Asterisk console that MusicOnHold() gets called and tcpdump shows the INVITE that first sets the RTP source to 0.0.0.0 then sets it to the IP of my Asterisk box. None of this is needed; Asterisk will stream MOH to ATA 2 all by itself, just by the fact that ATA 1 put ATA 2 on hold. You have over-complexified the setup :-) I'm not sure what you mean here. You do have to defind a MOH class for any channel not using default. I think the problem you have is that you have not indicated anywhere that you have set the MOH class for either channel to random. If you do not do that, it will try to use MOH class default. Make sure you test the default class with your 899 extension, or set the MOH class to random for the channels you are testing. HTH. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
[Asterisk-Users] actionID on manager events
Hello, all! I'm looking at the wiki page and info on the mailing list and I'm getting conflicting info... I am using the manager API from the telnet CLI and I am testing creating calls with it. I login with events: on and I can originate calls just fine. However, when I set ActionID on an Originate, I cannot see anywhere where that actionid carries into the Event output. But I found this on a post from January: Yes, ActionID is a value you can use when issuing a command. It there so that you can be sure you respond to your own responses not to someone else's or that you respond to an response instance in the correct way. In a multi-threaded app you might have several actions outstanding so you will need to know what response corresponds to which command. Which indicates that the actionid should be coming through. Is there perhaps some setting I'm missing? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
[Asterisk-Users] Snom 360 hold problem
Hello, I have a customer who said that their Snom 360 is joining calls by accident. The situation is that they had one call on the line and another call came in. She pressed the hold button on the phone and the two calls were joined together. I do have Call join on Xfer set to yes, but I thought that would only come into play when doing a transfer, not putting someone on hold. The phone is at firmware 4.1, and there are no new updates, so that shouldn't be it. Anyone else experience this behavior on the phones, or know if I need to turn off Call Join on Xfer? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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
Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...
We've deployed IP 300's, 500's, and 501's at customers and they work very well. On Thu, Aug 11, 2005 at 11:52:35AM -0700, Ing. Marlo R. Beltran G wrote: I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Snom 360 and firmware 4.0 problem
On Sat, Aug 06, 2005 at 08:50:58AM +0200, Christian Stredicke wrote: Please take a look at http://www.snom.com/howto40.html. We tried to make the upgrade procedure as smooth as possible, if you are having problems please tell us and we will try to make it more simple. For example, if you have a batch of phones give us an email and we will send you the files in one go. New phones dont need that upgrade procedure. It is only necessary when you are crossing the 4.0 version border. For example, all 320 already have the certificate installed already, so for 320 there is no need to go throught the procedure. For release notes for 4.0, please check out http://www.snom.com/snom360_release_notes.html. I found the license issue info on the website. So my question boils down to this: Does this explain why they would not register, or do I have to worry that there is some new setting which caused the problem? I do not want to go through the pain of upgrading the customer's phones (probably with a site visit) only to find that I have to downgrade them and go though it again with 4.1. Also, is there any way to tell the phone, *before a reboot* that I want it to update the firmware? I do most of my maintenance remotely, and I can tell the phones where to find new firmware and clicking Load will start the reboot process. However, I need a person there to press the Check button so that it will really update the firmware. Is there any way around this so that I can update the phone after-hours and remotely? Thank you... Not just for this answer, but for all the answers I get from this list! I've been working with asterisk for a bit over a year, though I do not know near as much as many of you. I try to chime in with answers when I can, but I have received much more info from this list than I have contributed. This list is a true shining example of how Open Source Software can work! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Friday, August 05, 2005 8:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Snom 360 and firmware 4.0 problem I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the error occurs on that one, too. The distributor's latest suggestion was to go to the newest firmware, 4.0. I did that on the new 360 (from the RMA) and with the same account settings as the one it was replacing, it could not register with *. Since I was in a pinch, I updated the firmware down to the latest below 4.0 and the phone works just fine. Does anyone with more knowledge than I know what might be going on? Maybe a new default setting in 4.0 that's breaking things? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] Snom 360 and firmware 4.0 problem
I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the error occurs on that one, too. The distributor's latest suggestion was to go to the newest firmware, 4.0. I did that on the new 360 (from the RMA) and with the same account settings as the one it was replacing, it could not register with *. Since I was in a pinch, I updated the firmware down to the latest below 4.0 and the phone works just fine. Does anyone with more knowledge than I know what might be going on? Maybe a new default setting in 4.0 that's breaking things? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] strange dial problem with polycom 501
I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above, if just typed into the phone, might end up: 9507291234. Other times the cursor might jump right back to the beginning of the number. This doesn't happen when they enter the number and the press dial, so it seems to be a digitmap problem. However, the digitmap is nearly the same as what I've used on IP-500s in the past. It is: [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T [Actually it was [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T -- I don't know where that space came from, but I'll take it out and test again today.] Are there any obvious problems with that digitmap? Anything else that I should take a look at? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] musiconhold in sip.conf
I have a PBX set up so that all the channels have musiconhold=classname. We use all SIP phones and SIP to the VoIP provider. All channels have that setting. I directly call SetMusicOnHold(classname) for all incoming calls, but outgoing calls should be set by the musiconhold= configuration parameter. However, I still have circumstances that will play the default MOH class, but I cannot narrow down the circumstances which cause the classname setting to be lost. I'm using asterisk 1.0.7, Polycom phones, and a SIP provider. I have confirmed that all sip.conf entries have musiconhold=classname set. Most users are using the Polycom hold and transfer capability, if that should make a difference... Any other things I should look for to find the cause of the loss of that setting? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] snom 360 audio garbled
On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote: You might try checking which codec is in use pre-hold and post-hold. On our Snom 190s, g726 always seems to sound garbled, and the call may be starting with one codec (like ulaw), then continuing with another (like g726) after being taken off of hold. Do you mean which codec asterisk thinks it's using or which codec the phone is using at a particular time? I know that sip show channels will list the codecs used on a channel by *, but how do I find out what codec the Snom is using on an ongoing call? Also, if the snom went to a different codec, wouldn't the audio be completely incomprehensible? What we get is garbled and we can hear what sound like syllables, but you cannot really make it out. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] snom 360 audio garbled
On Tue, Jul 19, 2005 at 02:17:07PM -0400, Scott wrote: Do you mean which codec asterisk thinks it's using or which codec the phone is using at a particular time? I know that sip show channels will list the codecs used on a channel by *, but how do I find out what codec the Snom is using on an ongoing call? Also, if the snom went to a different codec, wouldn't the audio be completely incomprehensible? What we get is garbled and we can hear what sound like syllables, but you cannot really make it out. Thank you. Yes, I mean running sip show channels both pre- and post-hold and seeing if the codec listed for the channel is different. If it is, try disallowing the second codec in sip.conf. Okay, I'll check that. Is there a way to disable all but the first codec from the GUI? I've not yet gotten to the mass-configuration yet... Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] snom 360 audio garbled
On Tue, Jul 19, 2005 at 02:17:07PM -0400, Scott wrote: Michael George wrote: On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote: You might try checking which codec is in use pre-hold and post-hold. On our Snom 190s, g726 always seems to sound garbled, and the call may be starting with one codec (like ulaw), then continuing with another (like g726) after being taken off of hold. Do you mean which codec asterisk thinks it's using or which codec the phone is using at a particular time? I know that sip show channels will list the codecs used on a channel by *, but how do I find out what codec the Snom is using on an ongoing call? Also, if the snom went to a different codec, wouldn't the audio be completely incomprehensible? What we get is garbled and we can hear what sound like syllables, but you cannot really make it out. Thank you. Yes, I mean running sip show channels both pre- and post-hold and seeing if the codec listed for the channel is different. If it is, try disallowing the second codec in sip.conf. According to the distributor, the lastest firmware revisions addressed this problem. I have now upgraded the firmware and I hope it helps. I'll see later this week. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] snom 360 audio garbled
I have a new snom 360 on an internal net to my * box. When putting a call on hold and taking it off, the audio will usually be broken and not understandable. Sometimes this happens on incoming calls and almost always on outgoing calls. Anyone run into this before? Thx! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] Polycoms and paging
I'm looking at deploying some Polycom 501's here, but one thing that still needs confirmation before I can move forward is global paging. I figure that I can couple polycom auto-answer (http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config) with this script: http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html However, that script was posted over a year ago. I'm hoping someone can confirm for me whether the info on these two pages will still work with * 1.0.7+ so I know what to expect to get paging to work. I'm not sure how much time investment my superiors will accept and while paging is not a make-it-or-break-it feature, they will be duly impressed if I can get it working. Thanks for any feedback anyone might have on this... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Snom 360 NOTIFY syntax
On Mon, Jul 11, 2005 at 01:16:08PM -0500, Patrick Friedel wrote: I'm rolling out an installation with snom 360s in the near future. Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I have the 360's set up to subscribe and notify for the line use lights, which works like a charm for interoffice calling (between the 360's, anyway. The IAXy, 200 and, softphone will be used by less phone dependant types) but what I can't figure out from the Wiki is if it's possible to have the ZAP lines notify for the outbound lines so we can see how many lines are in use. I am by no means an expert at this, but I did some experimentation and it appears that the NOTIFY will not get sent for the trunk lines, only for extensions. I also found that the SUBSCRIBE/NOTIFY sequence only works for SIP and ZAP, I couldn't get it to work for IAX2. I do not know why this would be, and it is possible I was doing something wrong, but for what it's worth, that's my experience so far. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] verbosity in log files
I have an installation where one of the users claims that they have had calls where they could hear audio and the other party could not. I went to /var/log/asterisk and looked at messages and full, but they didn't have much info in them. I have: full = notice,warning,error,debug,verbose in logger.conf, but the log file doesn't look anything like the verbose output I get from the CLI. Is there another setting somewhere that I need to turn on so that full is full? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] voicemail.conf overwritten
I'm trying to set up a base * configuration in a version control system. I have nearly all the system-specific configuration pulled out into a subdirectory so that /etc/asterisk is very generic and I can copy it into another system when I create it. The only stickler is voicemail.conf. Includes within voicemail.conf will work, but when a user changes their password, * cannot put the change into the included file and a restart would wipe out password changes. I tried using symlinks and hard links and it appears that when a user changes their VM password, the file is deleted and rewritten. Those semantics defeat both types of links. I will have to ponder this issue for a bit. If anyone has suggestions, please offer them. If I come up with a solution transparent to the revision control system (subversion), I will report back to the list. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Asterisk 1.0.7 on Gentoo
You haven't upgraded your kernel since you installed the zaptel package, did you? If you do: ls /lib/modules/kernel version/misc do you see ztdummy.ko in there? I have a system running 2.6.11-gentoo-r6 and zaptel 1.0.7 with no problems. However, I load the modules in /etc/modules.autoload.d/kernel-2.6 with: zaptel ztdummy and I have asterisk and zaptel running at default runlevel with asterisk depend()-ing on after zaptel. See how that compares to your configuration... On Fri, Jun 03, 2005 at 08:17:57AM -0400, Waldo Rubinstein wrote: It also fails. # /etc/init.d/zaptel start * Starting zaptel... Notice: Configuration file is /etc/zaptel.conf line 206: Unable to open master device '/dev/zap/ ctl' [ ok ] # lsmod Module Size Used by Any other ideas? Thanks, Waldo On Jun 3, 2005, at 12:56 AM, Dan Perik wrote: Waldo Rubinstein wrote: I installed Asterisk on Gentoo using emerge. At first, emerge tried installing version 0.9 but reading the wiki showed how to get the latest stable. I'm running Gentoo kernel 2.6.11-gentoo-r9. Asterisk seems to be working just fine, but I'm concerned that since I don't have any Digium hardware, I may need a timer source. When I executed emerge zaptel, it installed zaptel 1.0.7 as well. The problem is that I can't seem to be able to load ztdummy or any zaptel module. I'm running * on Gentoo. Just a shot in the dark here. Have you tried: /etc/init.d/zaptel start Then do your modprobe. Let us know what happens. - Dan ___ 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 ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] SIP_CODEC, reinvites, and changing codecs
I am wondering if the SIP protocol and its implementation in * allows for changing codecs mid-connection. I've seen some questions regarding this on the list, but I've not found any clear answers. I've also seen the SIP_CODEC variable, but it's not clear that it will change the codec on an existing call. Also, there are mentions of needing a reinvite to make the change, but most of the sample sip.conf contexts I've used for setting up our sip channels reccommend canreinvite=no. Does that preclude any change I might've had in changing codecs? Basically, what I have is polycom phones with 729 licenses and access to the VoIP provider which can do 729. Native bridging will not consume licenses, but accessing VM on either side will. Same with MOH. I have a license for each of the VoIP provider channels, so I'm not worried about changing their codec because I want the compression there always. They can just have licenses... I would like to have the polycoms connect initially with 729 which will eventually natively bridge for internal channel-to-channels calls, internal-to-trunk calls, and trunk-to-internal calls. But when an internal channel is accessing voicemail, I would like to change the codec in use from 729 to ulaw to release the license. Since these are on an internal 'net the bandwidth usage is not a big deal. Is this possible with canreinvite=no in the sip.conf entry for the polycoms? Can I achieve this with SetVar(SIP_CODEC, ulaw) in my dialplan before sending the internal call to VoiceMailMain()? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] iax went away
I just had a situation where I could not get calls from or out to one of my IAX2 boxes to another. The one which seemed to have a problem didn't show the server in its iax2 show registry list. I reloaded and the register showed up. Looking at the server, when I called the number I got the message: Jun 2 20:48:34 NOTICE[25542]: chan_iax2.c:2209 auto_congest: Auto-congesting call due to slow response Since I have qualify=yes, I suspected that slow response caused the server to take the other host off-line. However, things had been fine on the network most of the day. Just after the above message, I did an iax2 show peers and it listed that host as 26ms away. Not too bad... I did a reload on the server and after that all seems normal. So it appears that the server and client lost touch for a bit and apparently gave up trying to reach the other. I can see the server giving up if maybe the IP address changed or something, but why didn't the client keep trying to register? Did I maybe dork a config setting somewhere that it isn't retrying? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working
On Tue, May 31, 2005 at 12:06:55PM +0200, David Hajek wrote: Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Do you have the NAT Enable and NAT keepalive set to Yes on the Sipura? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Polycom phones, UNREACHABLE
On Sat, May 28, 2005 at 11:10:30AM -0400, Steve Totaro wrote: qualify = yes is what is causing the messages. You can assign a value rather than yes. like 1000 or something or you can remove the qualify statement alltogether. The message is just a warning. Eliminating the warning does not eliminate the lag problem. That's what I thought, that qualify=yes is only indicating the problem, but that the unreachability problem still exists. In a way, I like the warning because it tells me how often this happens. The bigger problem is *why* are the phones becoming unreachable? - Original Message - From: Michael George [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, May 27, 2005 11:26 PM Subject: [Asterisk-Users] Polycom phones, UNREACHABLE I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now UNREACHABLE! May 27 16:31:54 NOTICE[29728]: Peer 'Polycom1' is now REACHABLE! (Sometimes the first message says TOO LAGGED...) And as you can see these messages are quite a ways apart, not just a few seconds. I have read the archives and found some clues that decreased the frequency of the problem, but have not eliminated it. My configuration for the phones in sip.conf is: defaultexpirey=3600 ; this is required by our VoIP provider rather than 120 [Polycom_1] username=Polycom1 secret= type=friend canreinvite=no ; specifically recommended in archives nat=yes ; phone is behind a NAT qualify=yes ; I suspected this might help... host=dynamic dtmfmode=rfc2833 context=internal disallow=all allow=ulaw In the sip.cfg file for the phone on it's FTP server, I have set: -server.1.address to the public address of the server -voIpProt.SIP.outboundProxy.address to the public address of the server -nat.ip is not set, as the description doesn't make it look like I want to mess with it... -there are other possible settings in that file that might be helpful, but the descriptions are a bit thin in the manual... I want to deploy more of these phones, but if they are ducking off the server every so often, that makes them unreliable. Does anyone have any ideas what the problem might be? I think if I remove qualify=yes from sip.conf it will eliminate the warnings in the log, but I think the phone will still be unreachable for that time period and the problem is just less evident... Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Polycom phones, UNREACHABLE
Actually, it looks like I'm getting this problem on all my phones. When I was testing my phones, most worked pretty well with an occasional complaint from the Polycom. I've moved them now to a different location and the ISP must have different NAT translation going on that make it more difficult to penetrate the NAT. Am I right in guessing that even with qualify=no the problem would still be present (unreachable phones), but it wouldn't show up in the logs? Thanks! On Fri, May 27, 2005 at 11:26:35PM -0400, Michael George wrote: I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now UNREACHABLE! May 27 16:31:54 NOTICE[29728]: Peer 'Polycom1' is now REACHABLE! (Sometimes the first message says TOO LAGGED...) And as you can see these messages are quite a ways apart, not just a few seconds. I have read the archives and found some clues that decreased the frequency of the problem, but have not eliminated it. My configuration for the phones in sip.conf is: defaultexpirey=3600 ; this is required by our VoIP provider rather than 120 [Polycom_1] username=Polycom1 secret= type=friend canreinvite=no; specifically recommended in archives nat=yes ; phone is behind a NAT qualify=yes ; I suspected this might help... host=dynamic dtmfmode=rfc2833 context=internal disallow=all allow=ulaw In the sip.cfg file for the phone on it's FTP server, I have set: -server.1.address to the public address of the server -voIpProt.SIP.outboundProxy.address to the public address of the server -nat.ip is not set, as the description doesn't make it look like I want to mess with it... -there are other possible settings in that file that might be helpful, but the descriptions are a bit thin in the manual... I want to deploy more of these phones, but if they are ducking off the server every so often, that makes them unreliable. Does anyone have any ideas what the problem might be? I think if I remove qualify=yes from sip.conf it will eliminate the warnings in the log, but I think the phone will still be unreachable for that time period and the problem is just less evident... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] Polycom phones, UNREACHABLE
I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now UNREACHABLE! May 27 16:31:54 NOTICE[29728]: Peer 'Polycom1' is now REACHABLE! (Sometimes the first message says TOO LAGGED...) And as you can see these messages are quite a ways apart, not just a few seconds. I have read the archives and found some clues that decreased the frequency of the problem, but have not eliminated it. My configuration for the phones in sip.conf is: defaultexpirey=3600 ; this is required by our VoIP provider rather than 120 [Polycom_1] username=Polycom1 secret= type=friend canreinvite=no ; specifically recommended in archives nat=yes ; phone is behind a NAT qualify=yes ; I suspected this might help... host=dynamic dtmfmode=rfc2833 context=internal disallow=all allow=ulaw In the sip.cfg file for the phone on it's FTP server, I have set: -server.1.address to the public address of the server -voIpProt.SIP.outboundProxy.address to the public address of the server -nat.ip is not set, as the description doesn't make it look like I want to mess with it... -there are other possible settings in that file that might be helpful, but the descriptions are a bit thin in the manual... I want to deploy more of these phones, but if they are ducking off the server every so often, that makes them unreliable. Does anyone have any ideas what the problem might be? I think if I remove qualify=yes from sip.conf it will eliminate the warnings in the log, but I think the phone will still be unreachable for that time period and the problem is just less evident... Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] VoipSupply.com
On Fri, May 20, 2005 at 02:17:57PM -0600, Rich Adamson wrote: Guess that makes about 9 out of 10 happy customers... anyone want to make that 90 out of 100? We've only dealt with them a couple time. However, our first order consisted of a Vegastream ATA which was just flakey and a Snom 190 that had the handset quit working. We contacted Snom about the 190 and after a few days they said, phone must be broken, return to reseller. We contacted voipsupply about that and they said, You shouldn't contact Snom about a problem like that, you should call us first. Snom's turnaround will probably take a few days to resolve the problem and we'd like to have you back up and running sooner than that. They exchanged both units for us and their standard operating procesure is to advance ship replacements. So, in short, their customer support is EXCELLENT! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] What do you name yours
On Wed, May 11, 2005 at 05:40:57PM +0200, Dave Cotton wrote: On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. So far we have only installed singular machines for clients. So I name them palantir. I wanted a good name that I could reuse and it would make sense. So we have [EMAIL PROTECTED] and [EMAIL PROTECTED] and [EMAIL PROTECTED], etc... Seemed like a cool thought at the time... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Kphone--asterisk--Kphone
On Tue, May 10, 2005 at 12:01:17PM +0530, Sudhananda wrote: I am running asterisk on one linux PC and want to talk through this server using Kphone installed on 2 different PC's. These are the extra lines added to sip.conf and extensions.conf respectively. sip.conf [jitha] type=friend host=dynamic secret=jitha context=sip dtmfmode=inband [sudhananda] type=friend host=dynamic secret=sudhananda context=sip This is what I use for kphone and it works fine: [kphone] type=friend ; either friend (peer+user), peer or user host=dynamic ; we have a static but private IP address callerid=kphone 25 dtmfmode=inband ; either RFC2833 or INFO for the BudgeTone context=internal disallow=all ; need to disallow=all before we can use allow= allow=ulaw; Note: In user sections the order of codecs extensions.conf [sip] exten=1,1,Dial(SIP/jitha,20,tr) exten=2,1,Dial(SIP/sudhananda,20,tr) Both the Kphones got registered to the asterisk but when i dial the number it gives me the following log on asterisk Asterisk Ready. *CLI -- Registered SIP 'sudhananda' at 172.16.2.35 port 5060 expires 900 -- Executing Dial(SIP/sudhananda-aa77, SIP/jitha|20|tr) in new stack -- Called jitha -- SIP/jitha-f4bc is ringing -- SIP/jitha-f4bc answered SIP/sudhananda-aa77 -- Attempting native bridge of SIP/sudhananda-aa77 and SIP/jitha-f4bc I see no problems here yet. and one Kphone status is ringing and on other it is connected. how to solve this problem. You might want to check the codecs in use. Are they both on the local network? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Recommended Linux Dist. for Asterisk
On Wed, Apr 20, 2005 at 10:26:33PM -0500, Paul Shiflet wrote: I'm trying to find out what flavor of Linux people are choosing for their asterisk boxes. I have been using RH, but i'd like to try some different ones. It seems that RH is the common denominator in this rash of line noise problems. So some suggestions for what dist to use would be great. We use gentoo. Many people would not go that route, but we use that on our servers because when we are ready to update it, we can do so with less pain than with RHL/Fedora and SuSE, etc. The updates of the latter usually go okay, but there comes the time when we need to change major releases and that should be done with a clean reinstall. Now, with * you don't really need to do any changing as it will just sit there and work for the most part. However, since we have gentoo in many of our systems, we just stick with that. The ports in gentoo stay pretty current and it's worked fine for us. YMMV, and as I said above, gentoo is probably not the route for many who have little linux experience. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] * not send SIP Notify for IAX2 channel
I am trying to get * to send a SIP notify to my SIP phone when an IAX2 channel goes active. * (1.0.5) is accepting the Subscribe and sending the notify just fine with the SIP channels and Zap channels, but not this IAX2 channel. I have this in my context as the hint: exten = 200,hint,IAX2/[EMAIL PROTECTED] and the channel use to dial that extension is: IAX2/[EMAIL PROTECTED] (with no context or extension needed) When I press the key that is loaded with that hint, the channel is dialed (and then the light will stay lit, but that's not the result of the NOTIFY) correctly. If I dial the extension directly, the NOTIFY is not sent. However, when another channel dials the IAX2 channel, the NOTIFY is only sent for the other channel, not the IAX2 channel. Is there some reason that * wouldn't send a NOTIFY for an IAX2 channel? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Snom subscribe/notify problem
On Mon, Apr 18, 2005 at 12:09:35PM -0400, Mailing List wrote: Does an underscore work? Yes, and underscore seems to work fine. Thanks for the suggestion. On Mon, Apr 18, 2005 at 11:34:03AM -0400, Michael George wrote: I have a Snom-190 that I've successfully used on a * box with the LED's lighting up when a line goes active. I have moved it to another box, though, and I'm having trouble with it. It almost seems as though there is a limit to how long a sip channel name can be for the subscribe/notify to work right. If I have the following in sip.conf: -- snip -- and this in extensions.conf: -- snip -- and the snom is set to light up it's LEDs for extensions 200-203. The LED's work just find when I call the snom (SIP/snom), but the light for the grandstream will not light up (SIP/PewTest-grandstream). If I change the entries for the grandstream from PewTest-grandstream to grandstream, then the light will work for that line, too. If I change the entries for the snom from snom to PewTest-snom, then the snom light fails to work. I have run sip debug mode on the snom peer and * is not sending out the NOTIFY messages, so it does not appear to be an issue with the Snom. Is there some type of limit to the SIP SUBSCRIBE/NOTIFY stuff that only allows 8 character channel names? It appears that the hyphen (-) in the channel name is what is breaking things. If I take that out, all seems to work fine. Anyone know why that might be? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] Snom NOTIFY on IAX2 channel
I'm setting up the LED keys on a Snom 190 and it is working fine for my other SIP clients. However, one of the extensions is an IAX2 channel to another * server. It can be dialed like any other extension (x200) and it can dial into the system. However, * will not send a NOTIFY to my Snom when that extension goes active. Has anyone been able to do that? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] cannot dial two phones using zap
On Mon, Apr 18, 2005 at 10:02:48AM +0800, Eddie wrote: So the Panasonic extension dialed by Zap/3/206 command will ring and Zap/4/221 will not ring at all, even before ext 206 is picked up? Yes, exactly. Zap/4/221 won't ring at all. If you have two extensions numbered 211 212, why are you using 206 and 221 in your Dial command? 211 212 is plugged to asterisk, for dialing purpose. 206 221 is the extension I want to dial to. I would try this: 1. Make sure either extension will ring all by itself. Yes, they do ring all by itself. Okay, so we know that either one will work by itself. 2. Ring both at the same time, but put them in the other order in the Dial() command and see if that makes a difference. I've tried this: exten = 3,1,Dial(Zap/3/206,10) exten = 3,2,Wait(2) exten = 3,3,Dial(Zap/4/221,10) exten = 3,4,Hangup Zap/3/206 won't hangup / timeout. It just keep ringing and won't stop. :) What does the * log tell you? Go to the CLI, set verbose 3 and see what happens when you dial the above dialplan. 3. Rather than having: channel = 3,4 try channel = 3 channel = 4 just for fun. Tried this. No difference. I'm not surprised, I didn't think it would do anything... 4. I don't know much about that Panasonic PBX, but are you sure calling two lines at the exact same time isn't messing it up? Not sure. If I were you, I would try testing without the panasonic PBX to make sure that the FXOs and your zap settings are correct. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] Snom subscribe/notify problem
I have a Snom-190 that I've successfully used on a * box with the LED's lighting up when a line goes active. I have moved it to another box, though, and I'm having trouble with it. It almost seems as though there is a limit to how long a sip channel name can be for the subscribe/notify to work right. If I have the following in sip.conf: -- [snom] type=friend ; Friends place calls and receive calls context=PewTest-snom ; Context for incoming calls from this user host=dynamic ; This peer register with us callerid=Snom190 201 dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info disallow=all allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! [EMAIL PROTECTED] ; Mailbox(-es) for message waiting indicator accountcode=PewTest amaflags=documentation; default AMA flag [PewTest-grandstream] type=friend ; either friend (peer+user), peer or user callgroup=1 ; We are in caller groups 1,3,4 pickupgroup=1 ; We can do call pick-p for call group 1,3,4,5 context=PewTest-internal username=grandstream1 ; usually matches the [section] title callerid=grandstream 202 host=dynamic ; we have a static but private IP address canreinvite=yes ; allow RTP voice traffic to bypass Asterisk dtmfmode=info ; either RFC2833 or INFO for the BudgeTone outgoinglimit=1 ; disable callwaiting signal (2nd call to phone) incominglimit=1 ; permit only 1 outgoing call at a time [EMAIL PROTECTED] disallow=all ; need to disallow=all before we can use allow= allow=ulaw; Note: In user sections the order of codecs accountcode=PewTest amaflags=documentation; default AMA flag -- and this in extensions.conf: -- [PewTest-snom] ;include = PewTest-internal ; extensions for monitoring exten = 200,hint,SIP/PewTest-sipura1 exten = 201,hint,SIP/snom exten = 202,hint,SIP/PewTest-grandstream exten = 203,hint,SIP/PewTest-grandstream exten = 200,1,Dial(SIP/PewTest-sipura1) exten = 201,1,Dial(SIP/snom) exten = 202,1,Dial(SIP/PewTest-grandstream) exten = 203,1,Dial(SIP/PewTest-grandstream) -- and the snom is set to light up it's LEDs for extensions 200-203. The LED's work just find when I call the snom (SIP/snom), but the light for the grandstream will not light up (SIP/PewTest-grandstream). If I change the entries for the grandstream from PewTest-grandstream to grandstream, then the light will work for that line, too. If I change the entries for the snom from snom to PewTest-snom, then the snom light fails to work. I have run sip debug mode on the snom peer and * is not sending out the NOTIFY messages, so it does not appear to be an issue with the Snom. Is there some type of limit to the SIP SUBSCRIBE/NOTIFY stuff that only allows 8 character channel names? Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Snom subscribe/notify problem
On Mon, Apr 18, 2005 at 11:34:03AM -0400, Michael George wrote: I have a Snom-190 that I've successfully used on a * box with the LED's lighting up when a line goes active. I have moved it to another box, though, and I'm having trouble with it. It almost seems as though there is a limit to how long a sip channel name can be for the subscribe/notify to work right. If I have the following in sip.conf: -- snip -- and this in extensions.conf: -- snip -- and the snom is set to light up it's LEDs for extensions 200-203. The LED's work just find when I call the snom (SIP/snom), but the light for the grandstream will not light up (SIP/PewTest-grandstream). If I change the entries for the grandstream from PewTest-grandstream to grandstream, then the light will work for that line, too. If I change the entries for the snom from snom to PewTest-snom, then the snom light fails to work. I have run sip debug mode on the snom peer and * is not sending out the NOTIFY messages, so it does not appear to be an issue with the Snom. Is there some type of limit to the SIP SUBSCRIBE/NOTIFY stuff that only allows 8 character channel names? It appears that the hyphen (-) in the channel name is what is breaking things. If I take that out, all seems to work fine. Anyone know why that might be? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] SIP calls being lost frame from cahnnel error
I've got a rather heated client that's having random drops of calls, mostly on transfer from one extension to another. They got me a precise call that disappeared on them and I found this message right before the call ended: Didn't get a frame from channel is the I've seen some mention of this in the mailing list, but nothing that indicates difinitively what happens to cause the problem or how to fix it. In our situation, we have Polycom 300's connecting to an * server, so it's not low-grade harware. I'm running * 1.0.5. It's a 1.2GHz Athlon with VoIP and only about 3 lines in use, usually 1-2 at a time, so it shouldn't be overloading the box... Anyone have suggestions as to where to look for a solution? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] new install
With the 2.6 kernel, you can just load ztdummy and not worry about the USB controller. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] cannot dial two phones using zap
On Fri, Apr 15, 2005 at 11:04:43AM +0800, Eddie wrote: I do understand how Dial works, but Zap/4 hungup immediately before Zap/3 is answered. Zap/4 doesn't even rings. So the Panasonic extension dialed by Zap/3/206 command will ring and Zap/4/221 will not ring at all, even before ext 206 is picked up? Sorry I didn't mention about this earlier, 206 221 are extensions connected to a Panasonic KX-TD1232 pbx. I missed that in your zapata.conf snipped. I have two extensions 211 212 connected to my TDM400p FXO ports. If you have two extensions numbered 211 212, why are you using 206 and 221 in your Dial command? I would try this: 1. Make sure either extension will ring all by itself. 2. Ring both at the same time, but put them in the other order in the Dial() command and see if that makes a difference. 3. Rather than having: channel = 3,4 try channel = 3 channel = 4 just for fun. 4. I don't know much about that Panasonic PBX, but are you sure calling two lines at the exact same time isn't messing it up? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Cannot dial two phones at the same time
On Thu, Apr 14, 2005 at 10:06:37AM +0800, Eddie wrote: I cannot dial two phones using zap at the same time. One will ring but the other one hangs up. Are those phones on an FXS or through an FXO to a PSTN to an outside number? zapata.conf [channels] context=default signalling=fxs_ks immediate=no busydetect=yes callprogress=no echocancel=yes echocancelwhenbridged=yes usecallerid=yes usecallingpres=yes threewaycalling=yes transfer=yes callerid=Incoming 20941261 group=1 channel = 3,4 extensions.conf [internal] exten = 300,1,Dial(Zap/3/206Zap/4/221,15) exten = 300,2,Hangup CLI linux*CLI dial [EMAIL PROTECTED] -- Executing Dial(OSS/dsp, Zap/3/206Zap/4/221|15) in new stack If these are FXS channels, then they should be of the form: Zap/chan. The additional argument you have, the 206 and 221, should not be present, AFAIK. From the wiki: chanspec[c][d][rcadence][/phonenumber] ... phonenumber, if present, specifies which telephone number you wish to be connected with. Note that this makes sense only when you are dialing a telephone line (an FXO or PRI interface), not an internal extension. Within ^ the phone number, you may use the special modifier w to indicate a half-second pause. You might want to use this to wait for a dialtone or for a pause while dialing digits. You may also use the special modifier c to allow for clear channel connections between PRI ports. -- Called 3/206 -- Called 4/221 -- Zap/3-1 answered OSS/dsp -- Hungup 'Zap/4-1' Console call has been answered Please advice. Thanks. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] ACCOUNTCODE lost after DISA()
On Wed, Apr 13, 2005 at 05:10:06PM -0400, Michael George wrote:
I am working on my dialplan, and I have come across many cool uses of DISA()
internally to generate dailtone at specific places where I want it. Works
quite well.
However, now I'm adding stuff to the dialplan that requires me to use the
ACCOUNTCODE predefined variable. Once I call DISA(), the subsequent
operations have an empty string for ${ACCOUNTCODE}. That seems odd.
I've checked the wiki and mailing list, but I don't see anything which seem to
relate to it.
running: CVS-v1-0-02/15/05
I am surprised that I have not heard anything back on this. Perhaps my
subject wasn't very good.
In essence, I am finding that after a call to DISA(), the ACCOUNTCODE variable
has been nullified. Is that expected behavior? Is there any way around it?
Thank you.
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] DISA() and predefined ACCOUNTCODE variable
I am working on my dialplan, and I have come across many cool uses of DISA()
internally to generate dailtone at specific places where I want it. Works
quite well.
However, now I'm adding stuff to the dialplan that requires me to use the
ACCOUNTCODE predefined variable. Once I call DISA(), the subsequent
operations have an empty string for ${ACCOUNTCODE}. That seems odd.
I've checked the wiki and mailing list, but I don't see anything which seem to
relate to it.
running: CVS-v1-0-02/15/05
Thanks!
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] snom and hint priority
Follwing the information from the wiki
(http://www.voip-info.org/wiki-Asterisk+phone+snom) and the mailing list, I
have been able to get my Snom 190 to monitor extension states accurately.
I have noticed a couple oddities, however, that I am hoping I can get
explanation on so that I can know more about * and SIP:
- It appears that I cannot use variables in the hint priority exten lines.
So exten = 22,hint,Zap/2 will work fine, but (assuming Ext22 = Zap/2)
exten = 22,hint,${Ext${EXTEN}} will not. Why is that?
- It appears that the extension used with the hint must be the same as the
extension used to dial that channel. So if extension 22 will ring Zap/2,
then exten = 22,hint,Zap/2 will work, but exten = 222,hint,Zap/2 will
not. Why is that?
- If I am correct in the above, then there is no way for me to monitor a
channel that is not an extension. As an example, I have a TDM400 with 3 FXS
(Zap/1-3 on extensions 21-23) and 1 FXO (Zap/4) as well as a VoIP channel
for dialing out. I can monitor the states of the extensions with extension
entries like exten = 21,hint,Zap/1 but I cannot monitor the state of the
FXO with exten = 0,hint,Zap/4 because 0 is not the extension of Zap/4.
Indeed, Zap/4 has no extension. Is it not possible to monitor that line,
then?
Thank you very much!
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] Cisco's description of echo
We are having trouble with an installation that is getting a lot of echo on some calls. The installation is all SIP phones and they have a VoIP provider. When we call through the voip provider and into another of their customers (voip throughout) there is no echo problem. If we call in their landline, through the TDM400's FXO to one of the SIP phones, there is no echo problem. Sometimes when we dial from SIP -- Voip provider -- PSTN -- destination it is okay, but other times the echo is horrible. In trying to figure this out, I found this article at Cisco's site: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml#1041385 It claims that echo always comes from the far end of the connection. So if I hear echo, then the origin of the echo is in the equipment on the end of the line near the person to whom I'm talking. The description seems to make sense, but the zapata.conf setting for echo cancellation seem to also help echo on the near end of the connection. I have read about echo on the wiki and in the mailing list, but it almost always discusses it with respect to the digium cards, not SIP alone. Is the Cisco article accurate? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Digium support quality: Excellent
On Tue, Mar 22, 2005 at 05:58:47PM -0500, [EMAIL PROTECTED] wrote: I wanted to make sure that, in addition to my complaints, I make it very clear: Digium's support is excellent. The jury is still out on the usefulness of the TDM products. However, Digium has worked very hard to make sure that this issue is resolved. I actually got an e-mail from someone at Digium actually asking what they could do to make me happy! She even gave me alternatives to hopefully correct my problem! And she was patient and friendly! I nearly fell off my chair. If you have any doubts about buying Digium products, don't let lack of support stop you. They stand behind their products with both technical support and customer service. You can't really ask for more than that. I agree that they are eager to correct any issues that we have with the cards. The unfortunate thing is that the TDMs are so problematic. I'm not sure if it's due to inconsistencies in the hardware into which they are put or the cards themselves or what. I have not yet successfully put 2 TDM cards into a system (though I know others have) and I recenly had a problem where loading the TDM driver and starting * would cause the outgoing message to be played way too fast. I was told to try changing PCI slots (I haven't had a chance to do that yet), but since the TDM cannot share IRQs with anything else, changing slots might just put it into a conflict situation. This one could be sticky... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] audio frequency with wcfxs and K8t
Friday and Saturday I was wrestling with a VoIP system that was having very strange problems. It was playing the outgoing IVR audio at 2-5x faster than it should have been. I found that if I stopped asterisk, removed the wcfxs driver and installed the ztdummy driver, the audio would play fine. I tested this in and out several times and it always worked fine with ztdummy and never worked right with wcfxs. I cannot find any references on the 'net about such a problem. Anyone else run into this? Details: Asterisk 1.0.6 MSI K8T Master2 motherboard Single AMD Opteron installed on the motherboard (other socket empty) TDM card with a single FXO installed on it The system is working fine now (SIP in and out), but I want to put a PSTN line into the FXO port as a land-line fallback. I also figured the TDM card could be the timing device for meetme and such, but I think ztdummy will do just as well there. Anyone else run into this? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Providing a dialtone
DISA() will give a dialtone. You might be able to work that into your dialplan... On Thu, Mar 10, 2005 at 09:43:15AM +1100, Howard Lowndes wrote: On Thu, 2005-03-10 at 08:05, Martijn van Oosterhout wrote: Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option that allows them to dial a normal number, to also provide a normal dialtone. Should I just record one and use Background()? I have a similar problem in as much as I want to provide a Facility dialtone to a zap channel under certain situations (call forward active) in the same way a Stutter dialtone is sent to a zap phone when there is a message waiting. Providing dialtone to SIP phones is probably not possible - I guess it is very phone dependent. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] iax notransfer=no and Tt in Dial()
I've not heard anything about this from anyone. I'm taking that to mean that I'm unique in having this problem. I think I will upgrade to a newer version of * and try again. I will report back with more questions or the solution. Thanks. On Tue, Mar 01, 2005 at 09:18:00PM -0500, Michael George wrote: I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set notransfer=no in the iax.conf file at the office so that the office system can step out of the media path and save a hop. That also works fine. However, that does not allow me to transfer someone who called my home extension at the office to someone else at the office. I have put the T/t options in the dial() command as I should. However, the office * box will still transfer the call, stepping out of the media path and breaking my ability to do the intra-office transfer. According to what I find in teh mailing list archives, putting a T/t as an option to dial() will halt a possible transfer and keep the system in the media path. However, that doesn't seem to be the case. I ran asterisk -vvvr to watch the call being processed and I can see the DIAL(channel||T) be called and shortly thereafter it gives the Ready to transfer and then indicates the hangup while the other two * systems are handling the channel. So what I see happening is not what the docs and archives say should be happening. Is this a new feature, that notransfer=no trumps T/t in the dial() command? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] cvs stable and 1.0.5
I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] cvs stable and 1.0.5
On Wed, Mar 02, 2005 at 09:49:02AM -0500, Clay Reiche wrote: Are you sure you're not looking at the date? Oh, you are probably right. It is 1-0-03/01/05, so that's 1.0 as of 3/1/5, not 1.0.3. So it appears, then, that the cvs will only display 1.0 and the .x part is only relevant for the releases. I also noticed that it's not recommended that one use the CVS version (even of stable) if not watching the asterisk-cvs list. Maybe, then, it would be best for me to revert to using the releases. What is the opinion of the list? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Wednesday, March 02, 2005 7:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cvs stable and 1.0.5 I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] iax notransfer=no and Tt in Dial()
I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set notransfer=no in the iax.conf file at the office so that the office system can step out of the media path and save a hop. That also works fine. However, that does not allow me to transfer someone who called my home extension at the office to someone else at the office. I have put the T/t options in the dial() command as I should. However, the office * box will still transfer the call, stepping out of the media path and breaking my ability to do the intra-office transfer. According to what I find in teh mailing list archives, putting a T/t as an option to dial() will halt a possible transfer and keep the system in the media path. However, that doesn't seem to be the case. I ran asterisk -vvvr to watch the call being processed and I can see the DIAL(channel||T) be called and shortly thereafter it gives the Ready to transfer and then indicates the hangup while the other two * systems are handling the channel. So what I see happening is not what the docs and archives say should be happening. Is this a new feature, that notransfer=no trumps T/t in the dial() command? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] make of asterisk doesn't do anything...
I got this figured out. Turns out that the make would call mpg123 with a 'version option to see if it was compatible with *. However, I've been faking out mpg123 with my own script to send raw audio to the device rather than mp3. As a result, my infinite cat was being spewed out and the results were not what make had expected. Once I rectified that situation, all went just fine. On Tue, Feb 15, 2005 at 07:34:35AM -0500, Michael George wrote: I just got the latest update from the 1.0 CVS tree this morning. I was able to make the zaptel drivers just fine, but in the asterisk directory, make just sits there. This is under the 2.4 kernel on a SuSE system which has worked just fine until now. I'm making as root, so it's not likely a permission problem. According to top, grep and cat are running with grep sucking down a huge amount of processor time. I did a make clean before the make, but that didn't help anything. It is a slow machine, but I let it run for like 15m and it hasn't produced the first bit of output. Anyone run into this? Thanks for any advice... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] SIP echo on LAN
On Mon, Feb 21, 2005 at 08:42:33AM +, Julian J. M. wrote: Check your soundcard controls... maybe it's recording what you hear or PCM, thus sending it again to the other party. Are you saying that when using a sound card with your softphone the PCM should be set to 0? I never knew that... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] make of asterisk doesn't do anything...
I just got the latest update from the 1.0 CVS tree this morning. I was able to make the zaptel drivers just fine, but in the asterisk directory, make just sits there. This is under the 2.4 kernel on a SuSE system which has worked just fine until now. I'm making as root, so it's not likely a permission problem. According to top, grep and cat are running with grep sucking down a huge amount of processor time. I did a make clean before the make, but that didn't help anything. It is a slow machine, but I let it run for like 15m and it hasn't produced the first bit of output. Anyone run into this? Thanks for any advice... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] make of asterisk doesn't do anything...
On Tue, Feb 15, 2005 at 01:16:16PM +, Alistair Cunningham wrote:
Michael,
Someone may know a simple fix. If not, can you please install the
'strace' program, then run:
strace -f -o /tmp/strace.out make
This will run make, and log any system calls it makes to
/tmp/strace.out. When it hangs, take a look in that file. It may have
stopped on one system call, such as select() or poll(), or it may be
scrolling endlessly, repeating the same system calls over and over
again. If grep is using a lot of processor time, it's probably the
latter. Either way, please paste the last 20 or lines into an email, and
post it to this mailing list.
It did just go on and on apparently reading and writing files. It seems to be
complaining alot about unfinished reads and writes... Below are the last
significant 40 lines.
I have saved the output file, so if looking at the first occurrence of PID's
might help, I can look it up.
Thanks!
[pid 22436] ... read resumed
p\5\341\5I\5\367\4\0\6\237\6{\6\t\6`\4\322\2\34\3\256\003..., 24576) = 8192
[pid 22436] read(0, unfinished ...
[pid 22440] ... write resumed ) = 4096
[pid 22440] read(3,
\367\373\270\374a\375\231\375W\3759\375\312\375\r\376B..., 4096) = 4096
[pid 22440] write(1,
\367\373\270\374a\375\231\375W\3759\375\312\375\r\376B..., 4096) = 4096
[pid 22440] read(3,
a\5\376\5\3\6.\7\337\10k\7\7\5I\5j\6*\6~\4\352\2U\3\336..., 4096) = 4096
[pid 22440] write(1,
a\5\376\5\3\6.\7\337\10k\7\7\5I\5j\6*\6~\4\352\2U\3\336..., 4096 unfinished
...
[pid 22436] ... read resumed
\230\6c\2[\377p\376d\377\274\1*\4\204\1:\376\37\373\272..., 16384) = 8192
[pid 22436] read(0, unfinished ...
[pid 22440] ... write resumed ) = 4096
[pid 22440] read(3,
\v\10\254\n)\20\254\22\275\22\252\17%\n\4\t\245\5T\2#\3..., 4096) = 4096
[pid 22440] write(1,
\v\10\254\n)\20\254\22\275\22\252\17%\n\4\t\245\5T\2#\3..., 4096) = 4096
[pid 22440] read(3,
\226\7\326\3d\0\250\2~\7\6\v\245\f\251\v\224\0105\4\310..., 4096) = 4096
[pid 22440] write(1,
\226\7\326\3d\0\250\2~\7\6\v\245\f\251\v\224\0105\4\310..., 4096 unfinished
...
[pid 22436] ... read resumed
\367\373\270\374a\375\231\375W\3759\375\312\375\r\376B..., 8192) = 8192
[pid 22436] read(0, unfinished ...
[pid 22440] ... write resumed ) = 4096
[pid 22440] read(3,
Q\371u\373\310\371_\364\211\365+\365K\363\355\373\311\3..., 4096) = 4096
[pid 22440] write(1,
Q\371u\373\310\371_\364\211\365+\365K\363\355\373\311\3..., 4096) = 4096
[pid 22440] read(3,
\257\375r\374\340\375+\0j\0\324\377\30\0\242\0\360\0h\1..., 4096) = 4096
[pid 22440] write(1,
\257\375r\374\340\375+\0j\0\324\377\30\0\242\0\360\0h\1..., 4096 unfinished
...
[pid 22436] ... read resumed
\v\10\254\n)\20\254\22\275\22\252\17%\n\4\t\245\5T\2#\3..., 65536) = 8192
[pid 22436] read(0,
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] make of asterisk doesn't do anything...
On Tue, Feb 15, 2005 at 01:16:16PM +, Alistair Cunningham wrote: Michael George wrote: I just got the latest update from the 1.0 CVS tree this morning. I was able to make the zaptel drivers just fine, but in the asterisk directory, make just sits there. This is under the 2.4 kernel on a SuSE system which has worked just fine until now. I'm making as root, so it's not likely a permission problem. According to top, grep and cat are running with grep sucking down a huge amount of processor time. I did a make clean before the make, but that didn't help anything. It is a slow machine, but I let it run for like 15m and it hasn't produced the first bit of output. Anyone run into this? Thanks for any advice... I did a make -v and it looks like the make hangs when trying to make .depend: No implicit rule found for `all'. Considering target file `depend'. File `depend' does not exist. Considering target file `.depend'. File `.depend' does not exist. Finished prerequisites of target file `.depend'. Must remake target `.depend'. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 1 unreaped children. Got a SIGCHLD; 1 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 1 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 2 unreaped children. Got a SIGCHLD; 1 unreaped children. Putting child 0x080730e0 (.depend) PID 24062 on the chain. Live child 0x080730e0 (.depend) PID 24062 I presume no one else is having this trouble...? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Snom programmable leds / keys usage for pickup groups?
On Tue, Feb 08, 2005 at 01:47:57PM +0100, Remco Barende wrote: Would it be possible to use the programmable led+keys on the Snom phones to signal that there is an incoming call that is ringing a call group or pickup group? We use this on our existing PBX if for example the accounting dept. is out for lunch but nobody can hear their phones. This way you can see an incoming call (and we hate voicemail) :) I'm also curious about how configurable the Snom's buttons are. Can they be assigned, say SIP/1, SIP/2, SIP/3, etc and light up when that channel is in use? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Voicemail timeouts after 30sec's everytime.
On Mon, Feb 07, 2005 at 11:08:51PM -0500, Jon Radon wrote: Instead of hijacking the thread you could just look it up. (HINT: it's a feature in cvs) I'm using stable rather than CVS. I did look on voip-info and I searched the mailing list archives. If there's another place I could've looked before asking, I'd love to know it to save redundant questions here in the future. Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] kphone and *
I'm having trouble with kphone on our system. It's using ulaw on an internal network. No NAT. I had it working fine with very similar hardware (an old Dell Optiplex GX1) running as an LTSP terminal. But then I put the same sound card in an Optiplex G1. Kphone will answer the line fine when I call it (call coming from the * machine), but when we try to get kphone to dial, each GUI button-press takes like 1min to respond. When calling kphone, dtms tones send fine, but hangup and hold will take 1min. In teh * log, I get lots of chan_sip maximum retries errors. But even with them, eventually kphone will do what it's supposed to do. I read in the archives that changing bindaddr from 0.0.0.0 in sip.conf to the * server's IP address might help. However, that seemed to be more for NAT * problems not internal problems. Kphone isn't traversing a NAT to get to *. So I doubt that's the problem. Anyone have any suggestions what I could look into? It's kphone 4.0.5 and * 1.0.2... Thanks. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Voicemail timeouts after 30sec's everytime.
On Tue, 08 Feb 2005 13:46:26 +1100, David Uzzell [EMAIL PROTECTED] wrote: Brian Dingman wrote: This is just a guess, but try an Answer before sending it to VM. Hmm ok not sure what that would do but I am willing to try anything at the moment. Here is the incomming from Extensions.conf [default] exten = 61290071091,1,Wait,1 exten = 61290071091,n,Answer exten = 61290071091,n,DigitTimeout,3 exten = 61290071091,n,ResponseTimeout,5 exten = 61290071091,n,Dial(SIP/800,60) exten = 61290071091,n,Waitexten exten = 61290071091,n,Playback,voicemail/default/801/unavail exten = 61290071091,n,Voicemail,801 exten = 61290071091,n,Goto,t|1 What are the n priorities in the above? I thought the priorities had to be explicitly set on each exten line... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] TDM400 - channel out to lunch?
On Wed, Jan 26, 2005 at 06:11:31AM -0600, Rich Adamson wrote: For those of us that have had probems with the tdm dropping, it seems stopping *, stop and restart zaptel, restart * fixes what seems to be a software bug. No reboot necessary. If that doesn't fix the problem, then you might have a defective module. I found that I need to unload and reload the wcfxs module from the kernel and re-run ztcfg. Perhaps the former is no necessary, but it's in my script now. There was an issue with the first tdm cards shipped (ver e/f) where the first module slot had a problem. Those that received replacement cards found an added jumper wire on them suggesting a printed circuit board trace had been missed (or something like that). I have been having trouble with E/Fs (the H seems to be more stable), but it's not just with the first module. In my case it is the second one. And I initially had trouble because the FXO was on socket 1. Digium had me move it to socket 4 and that helped some. But only for a time. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Power Alarm Error - Help
On Sun, Jan 23, 2005 at 11:56:50AM -0600, Michael K. Rodriguez User wrote: I had a similar problem with power. I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up and asterisk would load but the T1s on the Quad T1 card failed to come up. I placed a loop on the card and still no change. Finally, I removed the UPS and the T1s came up. Do know if this will help you, but the T1 card seems to be delicate with power. I've been having similar trouble with one of my units. I put a UPS on the system and it seemed to get better, but how that module fails regularly. Incidentally, when a module dies and holds the circuit open, I can top asterisk, unload the kernel modules, reload them, run ztcfg, and restart * w/o restarting the system. However, it does require * to stop and that is annoying. I moved my problematic phone to a different fxs module and all seems fine. We have a similar problem with a TDM/FXS module in a different location. I've written digium support, but they are kinda slow in responding. On 1/23/05 10:31 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been getting the following message in Asterisk and it shuts Asterisk down, needing a reboot. Power alarm on Module 2 I have (1) TDM400P with (2) FXS (2) FXO cards (1) X100P card Any ideas? Since nobody answered, I'll guess something :) Did you plug the power on the TDM400P ? since you have FXS ports, you need to plug it in ___ 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 ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] TDM400 answers the line all the time!
On Mon, Jan 17, 2005 at 08:12:24AM -0600, Justin Carlson wrote: hi all, We have a TDM400 card with 4 wfo modules. now the modules load fine and when i start asterisk with on phone line connected it just starts spewing these messages: -- Starting simple switch on 'Zap/4-1' Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:51 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... but no one is calling. i have plugged in a analog phone and dialed out on this line before i used it for *. any help would be great. zapata.conf [trunkgroups] [channels] language=en context=routing group=1 immediate=no signalling=fxs_ks channel = 1-4 zaptel.conf fxsks=2-4 loadzone = us Is there a reason you have fxsks=2-4 in zaptel.conf rather than 1-4? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] TDM400 answers the line all the time!
On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote: no i was using line 1 for testing /w fxs module and i never changed it back does changing it back make a difference? On Fri, 2005-01-14 at 07:43 -0500, Michael George wrote: On Mon, Jan 17, 2005 at 08:12:24AM -0600, Justin Carlson wrote: hi all, We have a TDM400 card with 4 wfo modules. now the modules load fine and when i start asterisk with on phone line connected it just starts spewing these messages: -- Starting simple switch on 'Zap/4-1' Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:49 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:51 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... but no one is calling. i have plugged in a analog phone and dialed out on this line before i used it for *. any help would be great. zapata.conf [trunkgroups] [channels] language=en context=routing group=1 immediate=no signalling=fxs_ks channel = 1-4 zaptel.conf fxsks=2-4 loadzone = us Is there a reason you have fxsks=2-4 in zaptel.conf rather than 1-4? ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] agent with queues remain unavailable duringtransferred call
On Tue, Jan 04, 2005 at 08:05:16AM +0100, Florian Overkamp wrote: On Mon, Jan 03, 2005 at 07:10:59PM +0100, Florian Overkamp wrote: On Mon, 2005-01-03 at 17:38, Michael George wrote: How are you transferring the call? With channel facilities (e.g. hook flash) or with the * '#' transfer? Calls are transferred using channel facilities (SIP announced transfer) That's your problem. It's documented at voip-info.org, but I don't remember where. You need to use #-transfer so that * knows that the line is now available for another call. That should help. Hmm, while this doesn't solve my problem it does point me in the right direction. I've just learnt that this is fixable behaviour and it should be/will change in cvs. I'll see what I can find out on that. That would be great if they change it! However, in the meantime you can enable transferring in * (add T to the Dial() options) and use that, no? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] agent with queues remain unavailable during transferred call
On Mon, Jan 03, 2005 at 03:53:16PM +0100, Florian Overkamp wrote: Hi, I'm seeing something I'd like suggestions on: I have a queue with agents that log in using agentcallbacklogin. The extension that is logged in with is a Local channel. Now, if a call comes in to the queue and is handled by an agent (in our case using Cisco 7960 SIP phones) and transferred (attended) to another extension, the agent remains unavailable during the remains of the call. Using show agents gives this: 103 (TIC 3) logged in on MGCP/aaln/[EMAIL PROTECTED] talking to Zap/20-1 (musiconhold is 'default') As you can see, the Agent is shown with the transferred call, and is unavailable for new calls. However, the phone _is_ on hook and free. I am using a 1.0.2. version (bri-stuff rc2b) Any suggestions are welcome. How are you transferring the call? With channel facilities (e.g. hook flash) or with the * '#' transfer? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] agent with queues remain unavailable during transferred call
On Mon, Jan 03, 2005 at 07:10:59PM +0100, Florian Overkamp wrote: On Mon, 2005-01-03 at 17:38, Michael George wrote: How are you transferring the call? With channel facilities (e.g. hook flash) or with the * '#' transfer? Calls are transferred using channel facilities (SIP announced transfer) That's your problem. It's documented at voip-info.org, but I don't remember where. You need to use #-transfer so that * knows that the line is now available for another call. That should help. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] queuing questions
I'm working with * and queuing and while things are mostly-working, they don't work quite as well as the docs on the wiki indicate they should. Things like leavewhenempty, the h option to queue, stuff like that. I search the archives and it seems that very few of the queuing questions are answered on the list. I'm curious why that is. I figure there are X possibilities: 1. The info is on the 'net somewhere and we aren't looking at those places. 2. Queuing is not often used so not many people have experience with it. 3. The people who are using it and have it going are too busy to read the list regularly and therefore the information doesn't make it's way back here. 4. Getting queuing working is a litmus test for a True * professional and those who have it working are protecting the secrets. If the answer is 1, I'd love to know where to look because I want queuing working correctly and I'm just beating my head between the docs and reality and getting nowhere... Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] verbose setting changed?
Up until last night, I could run: asterisk -vvvr as root to connect to a running * session and have the verbosity set to 3. Last night, however, I updated to CVS-v1-0-12/29/04-16:47:20 and the behavior is different. Now the -v flags don't seem to make a difference, I have to issue: set verbose 3 to change verbosity. Is that a planned change? One nice thing is that I only have to issue that one time on a running session is seems and the verbosity is remembered. However, my nightly asterisk -rx restart gracefully resets the verbosity back to 0. Is there a settings file that I can set verbosity in? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] queueing question
I'm trying to set up queueing on my system and for the most part it works fine. However, I'm trying to give the user the ability to break out of the queue. Putting the H option into queue() doesn't seem to work. That seems to hang up from the extension that was being dialed and then after the delay start ringing them again. Looking at the mailing list archives, others seem to have had the same problem. One solution is to use the context option in queues.conf to indicate a context for the user to have entries checked against. I created [quotcon] in extensions.conf, but with an extension in there of '*', I have to enter the * 2x before it will interpret it against the context. And this time I do not have H in the Queue() call, but the first * will give me the Hangup... message in the verbose output, just as happened when I had the H option and no [qoutcon]. The second * will actually be interpreted against [qoutcon]. In short, the only time the * is interpreted against the context is when the phone (only 1 while testing) is hungup. I am using the same technique as a posting to the list in 11/2004 indicated works for him, so I suspect I have some setting wrong, but I cannot find one Thanks for any help anyone may have! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Ouch, part reset, quickly
On December 3, 2004 1:04 pm, Andrew Kohlsmith wrote: On December 3, 2004 12:43 pm, Steven Critchfield wrote: Is it possible that your PSU isn't up to the task? If you aren't running a 400 or 500 watt PSU, I would be suspect of the PSU. That error message was attributed to not getting enough power before they put a power plug on the board itself. Now you know you aren't getting strangled by the PCI bus, but it still might not be enough power if you PSU isn't up to snuff to hold the power stable and high enough. I fully believe that that error is incorrect. I have been running into these problems on rev E, F and H cards on all manner of systems, from P90 with a 450W power supply to the Supermicro server chassis I hvae upstairs with triple-redundant power. Hell I even put a 100MHz DSO on the +12V rail and there is nothing there, the voltage doesn't move more than a dozen or so mV from +12.00V. For some it may be a power issue, but I believe there is either a power *distribution* issue on the TDM4XXP carrier, an electrical error on the FXS modules or even some kind of driver issue. I just started getting this on my system. I've been running it over a month in this system with few problem like this and now they are happening regularly... Is there a list of possible remedies for it yet? Anyone heard from Digium about the problem? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
Re: [Asterisk-Users] Ouch, part reset, quickly
On Mon, Dec 20, 2004 at 08:31:15AM -0500, Michael George wrote: On December 3, 2004 1:04 pm, Andrew Kohlsmith wrote: On December 3, 2004 12:43 pm, Steven Critchfield wrote: Is it possible that your PSU isn't up to the task? If you aren't running a 400 or 500 watt PSU, I would be suspect of the PSU. That error message was attributed to not getting enough power before they put a power plug on the board itself. Now you know you aren't getting strangled by the PCI bus, but it still might not be enough power if you PSU isn't up to snuff to hold the power stable and high enough. I fully believe that that error is incorrect. I have been running into these problems on rev E, F and H cards on all manner of systems, from P90 with a 450W power supply to the Supermicro server chassis I hvae upstairs with triple-redundant power. Hell I even put a 100MHz DSO on the +12V rail and there is nothing there, the voltage doesn't move more than a dozen or so mV from +12.00V. For some it may be a power issue, but I believe there is either a power *distribution* issue on the TDM4XXP carrier, an electrical error on the FXS modules or even some kind of driver issue. I just started getting this on my system. I've been running it over a month in this system with few problem like this and now they are happening regularly... Is there a list of possible remedies for it yet? Anyone heard from Digium about the problem? One suggestion I got from digium was to load the wcfxs module with the lowpower=1 option: modprobe wcfxs lowpower=1 I'll try it and see if it helps. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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
[Asterisk-Users] send # with transfer enabled
Every so often we need to send the # dtmf tones but * interprets that as the initiation of a transfer. The best solution I've found so far is outlined at: http://lists.digium.com/pipermail/asterisk-users/2004-March/039501.html This disabled transfer for a call. I take this to mean that there is no way to send dtmf for # with transfer enabled? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] more then two wildcards in one machine
On Thu, Dec 09, 2004 at 12:01:54AM +0200, Shoval Tomer wrote: Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. As others have said, you need to make sure you aren't sharing IRQs with the Digium cards. One way to easily avoid it is to go with the PowerPC architecture, as many have suggested. I have also noticed that multi-processor motherboards have a boat-load of IRQ lines, also. If you want to stick with the intel architecture, you can go that route, too. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Restarting *
On Thu, Dec 02, 2004 at 09:50:51AM -0500, Ferguson, Michael wrote: G'Day All What do I type at the command line to stop and start * on a RedHat ES3 box? I think that would be (as root): asterisk -rx restart gracefully or asterisk -rx restart when convenient -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] dtmf tones during conversation
On Mon, Nov 22, 2004 at 10:07:51PM -0600, Henry Devito wrote: This is called talk off. Try to turn relaxdtmf off. I set relaxdtmf=no and the situation has improved, but it does still happen some of the time. It's infrequently enough that I could live with it, but if I can completely eliminate the problem, I'd like to. Should I expect to eliminate it? Could a cordless phone on the zap channel perhaps exacerbate the problem? Any other setting I can try? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Radon Sent: Monday, November 22, 2004 9:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] dtmf tones during conversation I run into this with my Sipuras.. Generally happens with female voices. I think the adapter just thinks the tone from their voice is a DTMF tone. Annoying. On Mon, 22 Nov 2004 21:57:34 -0500, Michael George [EMAIL PROTECTED] wrote: I have a * box running our house and on one extension we are getting spurious DMTF tones during conversations. It only happens on one of the 3 FXS ports and it's the one w/ a cordless phone on it. At first I thought someone was being careless and just hitting a button on the other end of the line, but it's happening too much for that... Has anyone run into this before? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] dtmf tones during conversation
On Mon, Nov 22, 2004 at 10:07:51PM -0600, Henry Devito wrote: This is called talk off. Try to turn relaxdtmf off. Okay, I'll try that. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Radon Sent: Monday, November 22, 2004 9:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] dtmf tones during conversation I run into this with my Sipuras.. Generally happens with female voices. I think the adapter just thinks the tone from their voice is a DTMF tone. Annoying. On Mon, 22 Nov 2004 21:57:34 -0500, Michael George [EMAIL PROTECTED] wrote: I have a * box running our house and on one extension we are getting spurious DMTF tones during conversations. It only happens on one of the 3 FXS ports and it's the one w/ a cordless phone on it. At first I thought someone was being careless and just hitting a button on the other end of the line, but it's happening too much for that... Has anyone run into this before? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] dtmf tones during conversation
I have a * box running our house and on one extension we are getting spurious DMTF tones during conversations. It only happens on one of the 3 FXS ports and it's the one w/ a cordless phone on it. At first I thought someone was being careless and just hitting a button on the other end of the line, but it's happening too much for that... Has anyone run into this before? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] DISA() context restrictions
On Thu, Nov 11, 2004 at 10:58:37AM -0600, Michael Greb wrote: On Thu, 11 Nov 2004 09:33:29 +0200 (SAST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Tue, 9 Nov 2004, Michael George wrote: The only difference to my extensions.conf file is that if I have: exten = s,2,DISA(no-password, disa) -- Executing DISA(IAX2/[EMAIL PROTECTED]/6, no-password| disa) in new stack Nov 9 19:50:33 DEBUG[14521]: app_disa.c:160 disa_exec: Context: disa Bet you its the space after the comma. Notice that the Context: disa has two spaces. So try DISA(no-password,disa) without the space and see if that helps. If it does, its obviously a bug, but you have a work-around at least. Steve I wouldn't really call that a bug, especially since I've seen cautions in several places against including spaces. It's just the way it is, one wouldn't include spaces in a CSV file, nor inbetween comma seperated values in the GECOS field in /etc/passwd, so why between arguments in the dial plan. No fault of Michael George of course, he didn't know that was the case before but now he does... I just wouldn't call it a bug. I agree, not necessarily a bug. It would be nice if the spaces could be there, but that's just how it is. Now I know and hopefully others will pick it up even more easily in teh archives. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] DISA() context restrictions
I'm working on using DISA() to allow us to call into our * box from the outside and grab an internal line to dial out on. Pretty standard stuff. I've got the 1.0.x CVS version of * built and when I specify a context to the call, the first DTMF sent will cause * to hang up the line. I suspect there might be restrictions on contexts which DISA() can examine, but I cannot find anything in the wiki or archives to suggest that. I have been able to utilize the default DISA() context, but that doesn't leave it free for the other uses of DISA() I have in mind. The only difference to my extensions.conf file is that if I have: exten = s,2,DISA(no-password, disa) then the next button press will hang up the line. If I have: exten = s,2,DISA(no-password) then the next button-press in resolved in the disa context. I would expect, though, that both would act the same way. I have tried with and w/o a passcode, but the results are the same. I've also checked show application disa to see if perhaps semantics have changed but the docs had not, but there seems to be no change from the info on the wiki. Looking at the logs, on a failed call I see: -- Accepting AUTHENTICATED call from 216.157.203.105, requested format = 2, actual format = 2 -- Executing NoOp(IAX2/[EMAIL PROTECTED]/6, internal-voip| s| 1| Michael George 206) in new stack -- Executing DISA(IAX2/[EMAIL PROTECTED]/6, no-password| disa) in new stack Nov 9 19:50:33 DEBUG[14521]: app_disa.c:160 disa_exec: Context: disa Nov 9 19:50:33 DEBUG[14521]: app_disa.c:165 disa_exec: DISA no-password login success Nov 9 19:50:33 DEBUG[14521]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Nov 9 19:50:33 DEBUG[14451]: chan_iax2.c:5307 socket_read: Ooh, voice format changed to 2 Nov 9 19:50:33 DEBUG[14521]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Nov 9 19:50:33 DEBUG[14521]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Nov 9 19:50:34 DEBUG[14521]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals *** Nov 9 19:50:34 DEBUG[14451]: chan_iax2.c:5489 socket_read: Immediately destroying 6, having received hangup == Spawn extension (internal-voip, s, 2) exited non-zero on 'IAX2/[EMAIL PROTECTED]/6' Nov 9 19:50:34 DEBUG[14521]: chan_iax2.c:2403 iax2_hangup: We're hanging up IAX2/[EMAIL PROTECTED]/6 now... Nov 9 19:50:34 DEBUG[14521]: chan_iax2.c:2412 iax2_hangup: Really destroying IAX2/[EMAIL PROTECTED]/6 now... -- Hungup 'IAX2/[EMAIL PROTECTED]/6' and on a successful call I see: Nov 9 19:52:09 DEBUG[14618]: app_disa.c:160 disa_exec: Context: disa Nov 9 19:52:09 DEBUG[14618]: app_disa.c:165 disa_exec: DISA no-password login success Nov 9 19:52:09 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Nov 9 19:52:10 DEBUG[14451]: chan_iax2.c:5307 socket_read: Ooh, voice format changed to 2 Nov 9 19:52:10 DEBUG[14618]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Nov 9 19:52:10 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Nov 9 19:52:12 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals *** Nov 9 19:52:12 WARNING[14618]: cdr.c:286 ast_cdr_init: CDR already initialized on 'IAX2/[EMAIL PROTECTED]/7' -- Executing NoOp(IAX2/[EMAIL PROTECTED]/7, disa context) in new stack -- Executing Playback(IAX2/[EMAIL PROTECTED]/7, tt-monkeysintro) in new stack Nov 9 19:52:12 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'tt-monkeysintro' (language 'en') Nov 9 19:52:13 DEBUG[14451]: chan_iax2.c:5793 socket_read: Sending VNAK -- Registered to '69.73.19.178', who sees us as 24.11.146.21:4569 Nov 9 19:52:14 DEBUG[14451]: chan_iax2.c:3753 raw_hangup: Raw Hangup 69.73.19.178:4569, src=6, dst=230 Nov 9 19:52:14 DEBUG[14451]: chan_iax2.c:3753 raw_hangup: Raw Hangup 69.73.19.178:4569, src=6, dst=230 Nov 9 19:52:14 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Nov 9 19:52:14 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals -- Executing Playback(IAX2/[EMAIL PROTECTED]/7, tt-allbusy) in new stack Nov 9 19:52:14 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'tt-allbusy' (language 'en') Nov 9 19:52:15 DEBUG[14451]: chan_iax2.c:5489 socket_read: Immediately destroying 7, having received hangup Nov 9 19:52:15 DEBUG[14618]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals == Spawn extension (disa, 8, 3) exited non-zero on 'IAX2/[EMAIL PROTECTED]/7' Nov 9 19:52:15 DEBUG[14618]: chan_iax2.c:2403 iax2_hangup: We're hanging up IAX2/[EMAIL PROTECTED]/7 now... Nov 9 19:52:15
Re: [Asterisk-Users] DISA() context restrictions
On Tue, Nov 09, 2004 at 06:01:20PM -0700, Michael Loftis wrote: Yeah, stop putting spaces in your args in your dialplan. in your example it's trying to look for the disa context, not the disa context. I know, hard to get used to, but the argument processors are not intelligent at this point. Oh my. I figured it might be something like that. I presume that applies for everwhere in the dialplan... I'll have to clean that out tomorrow. Thank you so much!! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] disa hangs up on me
I've been playing with this a bit. I've found that if I give a context to DISA(), the next DTMF I send it will cause the line to hang up. However, if I let it default to the [disa] context, it will sometimes process tone and extension correctly. Even if I specify the disa context, the same one it would default to, the next tone will cause a hangup. So I tried renaming my [internal] conext [disa] as a test, but that also failed. This indicates that either: 1. DISA() cannot handle jumping to a context that is defined after the call to DISA() (unlikely), or 2. there is some characteristic of my [internal] context that is causing problems. I will continue to work on it and report back to the list. If nothing else, what I learn will be in the archives. Thank you. On Fri, Oct 29, 2004 at 07:27:52AM -0400, Michael George wrote: I have confirmed that DISA is the culprit. If I remove DISA from the s exten, ti works as I would expect -- I can dial internal extensions after getting in via iax. DISA is an important part of the office dialplan, though, as it allows us to call in from outside and get an internal line to dial out. I turned on debugging and if I have a passcode present and I enter it followed by the # key, I can see app_disa.c:268 disa_exec: DISA on chan IAX2/[EMAIL PROTECTED]:4569/2 password is good and app_disa.c:276 disa_exec: Successful DISA log-in on chan IAX2/[EMAIL PROTECTED]:4569/2 messages, so I know I'm getting logged in. However, as soon as I hit another digit, I get: --- Oct 29 07:21:45 DEBUG[131080]: chan_iax2.c:5489 socket_read: Immediately destroying 2, having received hangup == Spawn extension (internal, s, 2) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569/2' Oct 29 07:21:45 DEBUG[262160]: chan_iax2.c:2403 iax2_hangup: We're hanging up IAX2/[EMAIL PROTECTED]:4569/2 now... Oct 29 07:21:45 DEBUG[262160]: chan_iax2.c:2412 iax2_hangup: Really destroying IAX2/[EMAIL PROTECTED]:4569/2 now... -- Hungup 'IAX2/[EMAIL PROTECTED]:4569/2' --- The digit I entered was 7. If I take DISA out of the loop I get no dialtone, but entering 773 will play sample sounds, as I would expect. Has the operation of DISA() changed, or maybe something else in * since the older CVS version I have that might be causing this? Thanks! On Thu, Oct 28, 2004 at 10:25:27PM -0400, Michael George wrote: I'm having a problem with DISA(). On my home system, I have the local extensions starting in [internal]. The s extension in [internal] has a NoOp() for debugging on s,1 and DISA(no-password,internal) at s,2. This allows me to return to internal,s,1 and get a dialtone again. Like after leaving voice mail or something. I have the same thing set up in the office, but that one doesn't work right. I've only been able to test it with my Grandstream so far and dialing in via IAX from home (so I can dial an extension from home and get plopped into [internal] at work). The DISA() call works just fine, I get the dialtone and all, but as soon as I send a button press, it hangs up on me. It doesn't go to the invalid extension or anything, I just get -- Executing DISA(IAX2/[EMAIL PROTECTED]:4569/1, no-password| internal) in new stack == Spawn extension (internal, s, 2) exited non-zero on 'IAX2/[EMAIL PROTECTED]:4569/1' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569/1' from the IAX connection and something similar on SIP. The main difference between the two is that the home (working) system is running CVS-HEAD-09/21/04 and the work system (not working) is running CVS-v1-0-10/28/04 (the latest, I beleive). Is there something changed in DISA that it won't work for me to loop back to my internal context and give a dialtone? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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 --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
