[Asterisk-Users] Polycom phone digitmap question
Hi, all I have Polycom SP300 phones. My extension range is 1xx, so I added corresponding entry to the digitmap. By some reason this does not affect on-hook dialing. If I have phone off-hook all is ok. dial extension 102 for example and it connects. if phone is off-hooh, however, I have to press DIAL or take it off hook before number is sent. Any ideas? Thanks, Rudolf P.S. Happens on both SIP 1.3 and 1.5 firmware of SP300 ___ 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] G729 with 2 channels
how to configure the g729 with 2 channels in iax.conf. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] PRI got event: HDLC Abort (6) on Primary, D-channel of span 1
I also experienced this problem and the first thing that really helped out was changing the timing in the span line of the zaptel.conf. Change it to look like this (see below) and see if it helps out. I got the error much less after doing this and eventually got rid of the error completely by removing my raid drives, installing an IDE drive, enabling DMA mode on the Hard Drive, enabling APCI in my kernel (also enabling my motherboard chipset in the kernel). After doing this I got 20x the Hard Drive write speed, no interrupts are shared, and the error is gone. span=1,0,0,esf,b8zs Message: 1 Date: Wed, 29 Jun 2005 07:13:29 -0600 From: Michael Blood [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I receive this error on the asterisk console and it is pretty much ALWAYS coming up. Sometimes there will be a break where it does not display. We had our PRI provider test the lines and they claim that there is no signalling problem. It doesn't matter if there are no calls or if there are 10 calls in progress the error is still displayed. I also get an annoying popping or clicking sound but that doesn't always correspond with this error coming up so it is likely a separate issue. I have loaded all modules by hand like below as someone suggested in a search for HDLC errors on the list. insmod zaptel insmod wct1xxp Unfortunately it did not help Has anyone run into this in the past? Michael ;zapata.conf switchtype=national context=incoming_eli_pri_1 signalling=pri_cpe group=1 channel = 1-11 bchan=1-11 dchan=24 ;zaptel.conf span=1,1,0,esf,b8zs bchan=1-11 dchan=24 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 29 07:09:07 NOTICE[3094]: chan_zap.c:7394 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 ___ 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] Multiple ISDN BRI Units with Asterisk using Bristuff zaphfc in NT mode?
Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? -- Greetings, Carl Andersson a.k.a Zaphod Beeblebrox -=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=- ___ 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] why $cdr{'CALLERID'} and $cdr{'DNID'} are empty in perl agi with asterisk manager
hello
i am using ast-rad-acc.pl from portaone connected with
asterisk manager.
my (%cdr) = @_;
$cdr{'CALLERID'},
$cdr{'DNID'},
these are empty
why these two variables are not working on hangup
any comments
thanks
Kamran Ahamd
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RE: [Asterisk-Users] Polycom phone digitmap question
As far as I know, the dialplan autodialer only works when the phone is off hook. This of course allows for nonstandard numbers to be dialed without regard to the digitmap. I, for example have lots of *XX numbers like *69 and *82, but if I wanted to dial *8 for a pickup I just dial *8 and then pickup the receiver. So, I guess in a way, its really a feature! ;) Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rudolf Ladyzhenskii |Sent: Friday, July 15, 2005 11:05 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Polycom phone digitmap question | | |Hi, all | |I have Polycom SP300 phones. My extension range is 1xx, so I added |corresponding entry to the digitmap. | |By some reason this does not affect on-hook dialing. If I have phone |off-hook all is ok. dial extension 102 for example and it |connects. if phone is off-hooh, however, I have to press DIAL |or take it off hook |before number is sent. | |Any ideas? | |Thanks, |Rudolf |P.S. Happens on both SIP 1.3 and 1.5 firmware of SP300 | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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
[Asterisk-Users] nathelper vs. asterisk
Hello, I'm currently using OpenSER as REGISTER server and Asterisk for the call routing. Do i need the OpenSER nathelper module if i want to offer (mostly) automatic NAT traversal to my users or does Asterisk have the same functionality? It seems that the nathelper module should be able to automatically traverse any NAT as long as the User-Agents use symmetric RTP. Further it is possible (in the ser.cfg) to automatically detect if the use of nathelper is needed for a specific call. Is this also possible with Asterisk? I found the options 'canreinvite' and 'nat' in the sip.conf, but I can't find any information about what behaviour the 'nat' option does change. Further I don't want to set 'canreinvite' globally to 'no' as I don't want to proxy the RTP stream if this isn't needed. Should I use Asterisk for this task - or is nathelper the better option? /gst signature.asc Description: This is a digitally signed message part ___ 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] Beginners question -- IAX
Hi, all Can someone point me to a good resource on IAX? I am trying to do two things at the moment: 1. I want to learn 2. I want to conenct MozPhone to my * (wiki does not have much on it) 3. I want to connect two * servers at different locations. I have looked at asterisk wiki and dis not find IAX stuff (may be I did not dig deep enough). Thanks a lot, Rudolf ___ 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] Server side call waiting for SIP
Has anyone implemented call waiting on the server side for calls to SIP phones? I.e. where only one call is delivered to the phone, and the called party hears a tone for subsequent calls, and they can press a key sequence to switch between them, the switching being done on Asterisk rather than the phone. On a related topic, if I were to implement it myself, is there a clean way to play a tone to an arbitrary channel from an AGI script? I could use the manager interface and redirect the call to a Playtones extension then back again, but a neater way would be good. -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.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] Multiple ISDN BRI Units with Asterisk using Bristuff zaphfc in NT mode?
Carl Andersson wrote: Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? Hi Carl, I just started yesterday afternoon with exactly the same setup so you are a bit ahead of me. If anyone answers you directly then please be kind enough to forward their comments to me. I have not even tried to sort out trunks, bristuff or anything yet but it might be worth pointing out that my initial problems were that, using an old pIII motherboard with pci graphics, network plus 2 HFC bri isdn cards, I ran out of IRQs. I had to lock down exclude the irq for the network card before the 2 ISDN cards woke up. I now have the network 1 ISDN card on their own IRQ and the graphics and 2nd ISDN card sharing an IRQ. Maybe this could be a similar problem for you? I'm using this HW with AAH 1.1 Keep in touch, Cheers, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ 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] Got SIP response 406 Not Acceptable back from 10.0.0.10???
Hi, What could cause: Got SIP response 406 Not Acceptable back from 10.0.0.10 10.0.0.10 = Hardware FXS And are there any probable solutions? Regards, Dave Walker ___ 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] RE: 2 asterisks connected but trying to bridge
On 16/07/05, Anton Krall [EMAIL PROTECTED] wrote: Also, both asterisks have notransfer?yes and I get this -- Attempting native bridge of IAX2/[EMAIL PROTECTED] and IAX2/voipjet-9 Why? Seems its not taking the notransfer into account. native bridge is not the same as transfer. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated
Thanks Peter. Any other takers on the list on this one ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: David Wilson [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 15, 2005 11:46 AM Subject: Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated On Fri, 15 Jul 2005, David Wilson wrote: Thanks for your reply. Would srx show ccmsgs 1 help ? I am not familiar with the Sirrix line of BRI cards. However, someone else on the list may be, or you may be able to diagnose the problem yourself. Peter ___ 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] Beginners question -- IAX
Hi As a general note: if you want to start a new thread, don't reply to an existing message: Write a new message. Otherwise your message will appear as a reply and be buried somewhere down a thread that nobody cares about. On Sat, Jul 16, 2005 at 08:27:56PM +1000, Rudolf Ladyzhenskii wrote: Hi, all Can someone point me to a good resource on IAX? http://voip-info.org/wiki-Asterisk I am trying to do two things at the moment: 1. I want to learn 2. I want to conenct MozPhone to my * (wiki does not have much on it) I'd try iaxcomm for a nice, simple and free soft-phone. MozPHone has been around for a while, but I still don't see it in the firefox addons site. Any idea why? 3. I want to connect two * servers at different locations. I have looked at asterisk wiki and dis not find IAX stuff (may be I did not dig deep enough). So now you'll have to be more specific. Do you have an Asterisk system? Have you tried doing the above? If so: what are the problems? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] channel.c:41:31: asterisk/transcap.h: No such file or directory problem
On Fri, Jul 15, 2005 at 09:52:19AM +0100, Angus Comber wrote: Hello I am trying to get Asterisk to work with the Junghanns Quad BRI ISDN card. I am progressing slowly! Problem I am now experiencing is as below. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\1.0.8-BRIstuffed-0.2.0-RC8h\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN-c -o channel.o channel.c channel.c:41:31: asterisk/transcap.h: No such file or directory asterisk/transcap.h is added by the bristuff patch. Are you sure it was properly applied? BTW: If you build it now, try 1.0.9 and bristuff RC8j . Alternatively, patch 1.0.8 yourself for the callerid issues. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] RE: 2 asterisks connected but trying to bridge
How can I disable that native bridge stuff? The scenario I have here is this. The main asterisk is behind a nat firewall and is routing port 4569 to that asterisk. The remote asterisk is also behind a nat and firewall. Both asterisk are connected thru an openvpn and they can see each other perfectly. Sip phones are local to the remote asterisk and they connect to it. That asterisk has a dialplan which routes any call to the main asterisk via IAX2. Weird thing happens, when a sip phone calls any number, the call is routed thru the remote asterisk to the main one, but I see warning messages on the remote asterisk (local to the sip phone which is where they are connecting to) about sip too many retries. Calls go thru the main asterisk and when answered you get the native transfer messages and audio is on one side only, the remote sip phone can heard the call but any phones connected to the main asterisk cant heard the remote sip phone. Why is happening here, why the warning about too many retries for the sips on the remote asterisk (which is local to them) and why am I getting just one way audio since both asterisk connect to each other via an openvpn with any firewall enabled? Can somebody help me out on this, its killing me, asterisk works great locally but Im having a very hard time making 2 asterisk work with each other with phones connected locally to them. Hope you can help me out Guys with some tips from people that have dodged this kind of problems. Thank you. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Peter Bowyer |Sent: Sábado, 16 de Julio de 2005 06:00 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] RE: 2 asterisks connected but |trying to bridge | |On 16/07/05, Anton Krall [EMAIL PROTECTED] wrote: | Also, both asterisks have notransfer?yes and I get this | |-- Attempting native bridge of IAX2/[EMAIL PROTECTED] and | IAX2/voipjet-9 | | Why? Seems its not taking the notransfer into account. | |native bridge is not the same as transfer. | |-- |Peter Bowyer |Email: [EMAIL PROTECTED] |Tel: +44 1296 768003 |VoIP: sip:[EMAIL PROTECTED] |___ |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
[Asterisk-Users] BT / X100P impedance matching
I understand that the X100P card is matched to a 600 ohm impedance but the UK BT phone system is not (I haven't been able to find much information on the impedance of the UK system). Has anyone come up with an easy way to match the impedance between the two so the X100P can work in the UK? Presumably a simple transformer won't do the job since it won't pass the DC components? -- - SteveXMPP/Jabber: [EMAIL PROTECTED]Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ 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] BT / X100P impedance matching
Steve, The X100P card works ok in UK (I have 3 at the moment). The only problem I encountered with it was when I had my SKY box connected to the same line. This caused random hangups. Apart from that the card works ok and the UK callerid patch is fine for detecting the BT ids. I hope this helps Vassilis At 13:11 16/07/2005, you wrote: I understand that the X100P card is matched to a 600 ohm impedance but the UK BT phone system is not (I haven't been able to find much information on the impedance of the UK system). Has anyone come up with an easy way to match the impedance between the two so the X100P can work in the UK? Presumably a simple transformer won't do the job since it won't pass the DC components? -- - SteveXMPP/Jabber: [EMAIL PROTECTED]Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ 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
[Asterisk-Users] Zap channel not hangingup
Hello, I am following up on a previous mail of the same subject at http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html In a nutshell I have connected my asterisk behind a Siemens HICOM 118E for a small call center application. The external PSTN calls will land in HICOM 118E and will get routed to 4 extensions which are connected to a TDM400P (REV E/F -- 4 FXO modules) I have configured a small IVR in * which are accessed by calling the said extensions. But in this setup when the caller hangs up Zap channel is not detecting it and goes to time out. A sample output is given at the end of the mail. I am also having echo problems. Do I have to make any additional settings to get this working? All my configurations are available at http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html I have been trying to get this working for quite some time and any help will be much appreciated. regards, raj ___ 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] VPN's
Sure, I have more than 18 tunnels to manage here, and the only blocking effects are thuse that I have volontary encoded . ;-) I believe that Peter has missed something in the VPN parmeters themselves or not correctly understood how are his IPtables onto this two IPSec secure gateway... Peter, could you post us the content of your /etc/ipsec.conf file ? We can take a look here and verify what is not good. Best Regards, Francois BERGERET, France. - Original Message - From: Shamsul Arefin [EMAIL PROTECTED] To: Francois BERGERET [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 15, 2005 11:48 PM Subject: Re: [Asterisk-Users] VPN's Hi, We use firewall and VPN togather to connect around 5 remote sites, and never encounter these problems. Make sure that port 10,000 and above mentioned in ur rtp conf files are opened in ur vpn and firewall. also when u connect from remote site don't use public ip use privte behind firewall. If still have problem send me more detail and i will be more then happy to sort this out . Regards Shamsul Arefin Saktek Broadband telephony experts On 7/16/05, Francois BERGERET [EMAIL PROTECTED] wrote: Hi men, You have some IP ports blocked ! I use SuperFreeSwan and I encounter no problem with this kind of configuration. Do you have open all ports on your IPsec gateways ? Think to have a look to your IPchains or any kind of firewall you are running in your IPSec gateway. I use shorewall and it is possible to miss some rules or to let pass few ports only for protections between sites. You must describe more your configurations to see what... Good luck ! Francois BERGERET, [EMAIL PROTECTED], France. - Original Message - From: Armin Schindler [EMAIL PROTECTED] To: Peter Osborne [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Sent: Friday, July 15, 2005 8:35 PM Subject: Re: [Asterisk-Users] VPN's On Fri, 15 Jul 2005, Peter Osborne wrote: Hi All, I'm using Asterisk for my PBX, I have a remote office that is connected by a VPN link. I am using Openswan on my side and a Linksys box on the remote side. I have a Polycom IP300 on the remote side configured with a static IP address. When I call the phone on the remote side, it rings and establishes the call fine. The problem I am having is that the remote side can hear the call find but the local side hears nothing. Because of the VPN there are no firwalls in the way. Does anyone have some ideas or atleast how I can track down the problem. I had the same problem with VPN using 'netscreen' (or a similar name) boxes. When I switched from SIP to IAX protocol, it worked perfectly. I think the SIP voice UDP packets are blocked somehow, but I didn't investigated it further. Armin ___ 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 -- Best Regards Shamsul Arefin Saktek , Broadband Telephony experts ___ 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] Beginners question -- IAX
http://www.google.com/search?q=asterisk+iax -Original Message- From: Rudolf Ladyzhenskii [mailto:[EMAIL PROTECTED] Sent: Saturday, July 16, 2005 5:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Beginners question -- IAX Hi, all Can someone point me to a good resource on IAX? I am trying to do two things at the moment: 1. I want to learn 2. I want to conenct MozPhone to my * (wiki does not have much on it) 3. I want to connect two * servers at different locations. I have looked at asterisk wiki and dis not find IAX stuff (may be I did not dig deep enough). Thanks a lot, Rudolf ___ 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] Vonage to IAX DID to IVR = Poor DTMF
Vonage Softphone service works with Asterisk. Search this list for more details. -Original Message- From: Michael Stearne [mailto:[EMAIL PROTECTED] Sent: Friday, July 15, 2005 10:21 PM Subject: Re: [Asterisk-Users] Vonage to IAX DID to IVR = Poor DTMF Does Vonage work with Asterisk? How much is this type of plan from Vonage? Thanks, Michael ___ 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 macro?
For our hotel application, we don't want to have to write 50 voicemail entries, is there a way to do a voicemail macro in the same way as a standard extension macro? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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 management
For our hospitality system, voicemail management is an issue. I looked at vmail.cgi and it works for the user, but I need a higher level management capabikity, i.e., flush all email from extensions 1XX (Apartment1) when a guest checks out. Is there anything like that or does anyone want to work on this for me? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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] howto on ISDN HFC cards with AAH v1.1
Hi, Can anyone please point me in a direction as to how to set up these 2 pci cards with AAH 1.1? I have (am still) googling left, right center - but haven't found a definitive guide yet. The centos based setup lacks any of the tools I know (insmod, modprobe ) so it is time consuming just to even dig around the AAH box. There are no zaptel.conf files and on it goes. A shortcut pointer would be great. TIA, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ 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] [EMAIL PROTECTED] and Cisco 7910
[EMAIL PROTECTED] takes only 15 minutes to install in a Xeon 2.8. However downloading 700mb ISO file could take all night. But I guess that it worth it, because it is very easy to manage, however I can not make my Cisco 7910 work. --- Sergio Chersovani [EMAIL PROTECTED] wrote: Javier Chia ha scritto: Ok, thank you. It is strange that no body have installed any cisco sccp phone in [EMAIL PROTECTED] yep they are all using sip I guess. Sergio ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Paging (I know, AGAIN)
Hey everybody,
I've been trying to recreate a paging unit that we have in house that
basically, when a user calls extension 44, it records their message.
When they hang-up, it plays a notification tone and then plays back the
message. I thought this should be easy, I have a sound card in the
Asterisk box, I have chan_oss loaded and working, I planned on hooking
the sound card up to the aux port on the Bogen. The problem is I can't
get it to continue beyond the first sound file. Relevant portions of
the dialplan below:
In my globals:
PAGING=Console/DSP
I'm running Asterisk HEAD as of July 16, 2005
[livonia-page]
; Check to see if paging is in use
; If active = YES, goto line 9, else continue on to 3
exten = s,1,Set(active=${DB(paging/active)})
exten = s,2,GotoIf($[${active} = YES]?9:3)
; Set paging/active to YES
exten = s,3,Set(DB(paging/active)=YES)
; Log paging to console
exten = s,4,NoOP(Paging *Livonia*)
; Begin record (No longer then 30 seconds)
exten = s,5,Record(paging:gsm||30)
; Play stutter tone
exten = s,6,Dial(${PAGING}||A(local/stutter)g)
; Play recorded paging message
exten = s,7,Dial(${PAGING}||A(paging)g)
; Set paging/active to NO
exten = s,8,Set(DB(paging/active)=NO)
; Hang up
exten = s,9,Hangup()
I've used the g option because the archives said to, but g is for when
the 'called' party has hung up. In this case, the called party is the
console and it doesn't hang up. I've tried the h extension, but it
doesn't seem to go beyond h,1. Trying h,2 h,3 doesn't work.
Any suggestions would be appreciated.
Doug
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RE: [Asterisk-Users] VM Outcall: Rube Goldberg Edition
I have been trying to get this to work. I monitor the spool directory
and no call file is created. Am I missing something here?
My Config
Voicemail.conf
[general]
externnotify=/usr/local/bin/vm-notify.pl
[root@ root]# ls -lat /usr/local/bin/vm-notify.pl
-rwxrwxrwx1 root root 2993 Jul 13 21:37
/usr/local/bin/vm-notify.pl
2203= ,Kevin
Test,[EMAIL PROTECTED],,|tz=eastern|notify=12035551212|attach=yes|saycid=no|di
alout=internal|callback=internal|review=yes|operator=yes|envelope=no|not
ify=12034878966
2204 = 2204,Toni Hawkins,,,tz=central,notify=12035551212
[root@ root]# cat /usr/local/bin/vm-notify.pl
#!/usr/bin/perl -w
use Fcntl;
use Fcntl :flock;
$dial_context=ctdialing;
($vm_box, $vm_context) = $ARGV[1] =~/(.*)\@(.*)/; $current_vm_context =
;
if(!sysopen($vm_conf_file_handle, /etc/asterisk/voicemail.conf,
O_RDONLY)) {
printf(Cannot open /etc/asterisk/voicemail.conf!\n);
exit(1);
}
while($vm_conf_line = $vm_conf_file_handle) {
chomp($vm_conf_line);
if((substr($vm_conf_line,0,1) eq ;) || (length($vm_conf_line) ==
0)) {
next;
}
($tmp_vm_context) = $vm_conf_line =~ /\[(.*)\]/;
if(defined($tmp_vm_context)) {
if($current_vm_context ne ) {
exit(0);
}
if($tmp_vm_context eq $vm_context) {
$current_vm_context = $vm_context;
next;
}
} else {
if($current_vm_context eq $vm_context) {
($tmp_vm_box) = $vm_conf_line =~ /(\d+)/;
if($tmp_vm_box eq $vm_box) {
($dial_dest) = $vm_conf_line =~ /.*notify=(\d+)/;
if(!defined($dial_dest)) {
exit(0);
}
close($vm_conf_file_handle);
# If there's already a .call file for this mailbox then don't
do anything.
# If there isn't already a .call file then create it.
#$call_file_name = /tmp/ . $vm_box . .call;
$call_file_name = /var/spool/asterisk/outgoing/ . $vm_box .
.call;
if(!sysopen($call_file_handle, $call_file_name,
O_WRONLY|O_CREAT|O_EXCL)) {
exit(0);
}
flock($call_file_handle, LOCK_EX);
# Set the access and modification times to be 10 years in the
future so
# Asterisk will ignore this file while we are doing stuff with
it.
$long_time = time() + (10 * 365 * 24 * 60 * 60);
utime($long_time, $long_time, $call_file_name);
srand;
$call_delay=300 + rand(120);
# Build our .call file.
printf($call_file_handle Channel:
Local/[EMAIL PROTECTED], $vm_box, $vm_context, $dial_dest,
$dial_context);
printf($call_file_handle WaitTime: 30\n);
printf($call_file_handle RetryTime: %i\n, 60 + rand(5));
printf($call_file_handle MaxRetries: 12\n);
printf($call_file_handle Context: vm-notify\n);
printf($call_file_handle Extension: s\n);
printf($call_file_handle Priority: 1\n);
printf($call_file_handle Callerid: Voicemail Notify
\9852463509\\n);
printf($call_file_handle SetVar: VM_BOX=%s\n, $vm_box);
printf($call_file_handle SetVar: VM_CONTEXT=%s\n,
$vm_context);
# Unlock and close the file.
flock($call_file_handle, LOCK_UN);
close($call_file_handle);
# Set the access and modification times to be 10 mins in the
future so
#Asterisk will delay for 10 mins before processing this .call
file
$short_time = time() + $call_delay;
utime($short_time, $short_time, $call_file_name);
exit(0);
}
} else {
next;
}
}
}
[EMAIL PROTECTED] root]#
-Original Message-
From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED]
Sent: Friday, July 15, 2005 12:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VM Outcall: Rube Goldberg Edition
The script will create a file in /var/spool/asterisk/outgoing
That is the file that makes Asterisk make the call. This this file
exist when there should be a pending call? Also make sure your
externnotify= is set to the full path of the script.
Kevin wrote:
Thanks for the update. I had made that assumption after looking at
the
script but checked as I can't seem to get it to call. I added the
variable to the general section, created the script, made it
executable
and no call. I wait the 10 minutes and monitor the asterisk and
system
messages log. Is there any way to monitor the script or perl log to
see
what's going wrong?
-Original Message-
From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED]
Sent: Friday, July 15, 2005 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VM Outcall: Rube Goldberg Edition
Kevin wrote:
Is the pager filed in the vm config still for the outcall destination
or
where do you specify the number to call for the outcall?
Sorry.
You use notify= option in voicemail.conf:
3532 = 8711,Toni
[Asterisk-Users] Asterisk International Carrier Buildout - Create our own International networks for BEST pricing!
Asterisk Users, I am reposting to the Asterisk-Users list what I saw on the Asterisk-Biz list by Mr. Jeff Grammer, GOD BLESS HIM! I am in the Hamptons today, trying to whew a client, and he took a look at my Level3 Partner pricing, and laughed, as his rates were better than mine!!! To top that off, I am colocated within the Level3 datacenter, he has no computers, and gets better rates. So, Mr. Grammer, (read his below post to Asterisk-biz), hit the nail right on the head, when he posted that we should start our OWN Asterisk International Carriers buildout as we can CREATE better pricing for OURSELVES, as opposed to all the others Please provide insights, directions as to EXACTLY how we can attain, using Asterisk, our own International Networks. I am interested in placing Asterisk boxes in South America, India, and other places. When the call comes into my Asterisk box in Chicago, and the caller wants Mexico or India, I shoot the call directly to the my other Asterisk boxes in those respective countries. But we would still have to make arrangements, (Correct me if I am wrong), to have the call carried over the local telecom network in that respective country. Or switched to TDM for PSTN landlines. Please someone assist as this can be done. We need better pricing, and must create that ourselves. Martin O'Shield [EMAIL PROTECTED] 1-877-238-5956 Message: 5 Date: Fri, 15 Jul 2005 17:04:45 -0400 From: Jeff Grammer [EMAIL PROTECTED] Subject: [Asterisk-biz] List of US accessible IAX servers? To: 'Commercial and Business-Oriented Asterisk Discussion' asterisk-biz@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII First, if this is the wrong mail list, please ignore this post. I have built my own Asterisk server and would like to participate in any sort of nationwide effort to build an alternative to LD carriers. I only have a couple of B1 (1FB) lines to use right now, but if I could learn how to make money terminating my Asterisk server in a network of other Asterisk servers I could upgrade to a PRI and provide access to my local calling area bypassing the LD carriers. But honestly, I don't know where to start. (or even really how to make money doing that) Is there a list of US freely accessible IAX servers anywhere that I could look to join as part of their network? Thank you, Jeff Grammer __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Any Ideas??? 3rd time posting = Sipura SIP Phones Multi-Line Appearance... How to use? |-----WAS---- [Asterisk-Users] NEWBIE Question: Asterisk with multiline/button phones
Still looking for some direction with this subject: I think the term is called multi-line appearance Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to indicate someone is using 'line 1' and they should NOT be able to pick up 'line 1' so long as that 'line1' is in use by another phone. I'd like this to work in a SIP only environment. We don't have actual CO lines but have several SIP accounts being used like CO lines... Is there a way to make these phones do this?? This is a common feature on just about any conventional phone system 'line seize' as it may be called to some... How the heck do you do it with sip? and does Asterisk do this readily? Thanks! Steve This is a very newb. question. Been using asterisk very happily now for several months and am considering getting some of those really 'cool' multi-button phones with LEDs and buttons. It's unclear to me if it is a straightforward task to actually setup a multiline 'presence' on the phones where the LED's light up when someone picks up a 'line' or is using a 'line' or puts a 'line' on hold or park and then would like to pick it up from another phone just by pushing the 'line #3' or 'line #4' button that is on hold and lit/flashing. Is this something that Asterisk actually does with ease? Or is it this a really complicated thing to accomplish setup? In particular in a sip only environment... no actual phone PSTN (pots) 'line's involved but with multiple SIP voip accounts to work like 'lines' with real PSTN phone numbers. We have several VOIP SIP accounts. Thanks take care! Steve ___ 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] Memory leak in asterisk CVS
Hi, My Asterisk CVS is apparently not doing much (other than keeping SIP IAX2 registrations alive and doing some ZAP calls (without echo-cancellation), but slowly the memory is filling up, so much so that 100m virtual memory is used up within 12 hours and I have to restart the asterisk application every 48 hours to make sure I have enough memory... How can I help resolve this problem? Problem occurs on both Sangoma and Digium installed systems. Fedora Core 3 and Centos 4.1 don't make a difference either. My version is Asterisk CVS-HEAD built on a i686 running Linux on 2005-07-11 16:29:02 I have removed the mailbox entries in my sip.conf which greatly reduced this problem. So, I suspect it may be in the sip or iax channel application. I also run quite a bit of agi scripts but none of them were alive when these memory-usage increases as shown below over a 1 minute interval with only 4 zap channels alive (2 calls) occured: ps -AF output... using this script: n=1;while [ 1 ]; do i=`ps -AF|grep ast|grep sbin|grep -v grep`; m=`echo $i|cut -f 6 -d\ `;if [ `echo $m` -ne `echo $n` ]; then echo $i; n=`echo $m`;fi;done root 15875 26881 0 15727 46240 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46248 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46256 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46268 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46280 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46288 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp Hope we can fix this somehow. Walter Klomp Singapore. ___ 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] Hangup Detection with busydetect
My telco doesn't provide Disconnect Supervision or Polarity Change. So I figured I have to detect hangups with busydetect=yes in zapata.conf. I tested it. When the telco sends a busy tone * detects it and hangsup. So far so good. The problem is the telco doesn't always send a busy after remote hangup. Most of the time it sends a congestion tone. I am guessing these tones from what I read on indications.conf. diitdiitdiit for busy diit diit diit diit for congestion busy = 450/500,0/500 congestion = 450/200,0/200,450/200,0/200,450/200,0/200,450/600,0/200 Looks like I have the correct setting for my country in indications.conf, verified it with ITU tones document. So at this point I figure I need to somehow make * detect both busy and congestion as same and hangup. I tried different BUSYDETECT algorithms, poked around at source code. Couldn't figure it out. Just to test what happens, I tried to change the tones for busy and replace it with the tones for congestion in indications.conf. To my surprise * continued to detect the old busy tone correctly and ignored the new tones I put in. I did the same in zaptel/zonedata.c and still * continues to detect the old busy tone correctly and ignores the new one I put in. So at this point I am totally confused. I don't even know where * gets the information about the tones. I am using CVS-HEAD as of today. Thank you -- Mehmet ___ 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] InfoWeek Article on VOIP
Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] FreeBSD 5.4 (Asterisk 1.0.9) compile error
Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.0.9/channels' gmake: *** [subdirs] Error 1 *** Error code 2 Anyone got any ideas? Mark ___ 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] FreeBSD 5.4 (Asterisk 1.0.9) compile error
Did you do a make clean? I just, as in 1 hour ago, successfully installed 1.0.9 using the port on FreeBSD. Darren Wiebe [EMAIL PROTECTED] Mark Ackroyd wrote: Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.0.9/channels' gmake: *** [subdirs] Error 1 *** Error code 2 Anyone got any ideas? Mark ___ 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
Re: [Asterisk-Users] InfoWeek Article on VOIP
Michael Graves wrote: Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves How do you see an ATA as different from and IP hardphone? As far as I can tell having the phone and ATA integrated isn't all THAT desirable, but that's me, I like to be able to choose the features on my phone and be able to connect it to the net... But that's just me. ___ 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] InfoWeek Article on VOIP
I agree with you but not 100% with them. An IP to Ip call on an ATA flat out is better . Now don't even get me started about cellular. My Service dosen't drop calls in the middle of conversations. VoIP is a notch better than Cellular. Michael Graves wrote: Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
Re: [Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910
7910 works fine wiz asterisk but you can not transfer calls, for that reason I will sell mine if somebody is interested... jl 2005/7/16, Javier Chia [EMAIL PROTECTED]: [EMAIL PROTECTED] takes only 15 minutes to install in a Xeon 2.8. However downloading 700mb ISO file could take all night. But I guess that it worth it, because it is very easy to manage, however I can not make my Cisco 7910 work. --- Sergio Chersovani [EMAIL PROTECTED] wrote: Javier Chia ha scritto: Ok, thank you. It is strange that no body have installed any cisco sccp phone in [EMAIL PROTECTED] yep they are all using sip I guess. Sergio ___ 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 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] arrgg! www.voip-info.org down again (or too busy)
On Friday 15 July 2005 16:54, Peter Osborne wrote: You can alway use google's cache. Use site:www.voip-info.org when searching or type the full URL into google and click on the cached version. Pete On 15 July 2005 4:36 pm, Damon Estep wrote: Does anyone have a mirror of this running? Yes, it's annoying as hell. A few times I've been close to make a mirror for myself so that I can access it reliably. Not quite the same to go through google. I just have to figure out what it would take set one up... -- List Manager Network Voice Comunications, Inc. netwvcom.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] Bridging two FXO cards (X101P) problem
Hi the list :-) Wondering why I can't bridge two X101P FXO cards to forward an external call from a first X101P to another analog telephone outside my house throught a seconf X101P. [VACATION] exten = s,1,Answer exten = s,2,Dial(Zap/3/ww0161417888),120 exten = s,3,Voicemail(u1001) exten = s,4,Hangup exten = s,104,Voicemail(b1001) exten = s,105,Hangup I temps to do that to avoid missing calls during my summer vacations. Numbering is ok when receiving a call, but no sound is heard or only few peak distorsions in background if I speak loud in the phone. Normal calls are running well, I use this Asterisk box every day and this FXO cards are ok. What have I missed ? Thanks for any help. Best Regards, Francois BERGERET, France. ___ 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] [ANNOUNCE] chan_capi-cm-0.5.4 release
Hi all, on sourceforge.net I added the fixup release 0.5.4 of chan_capi-cm driver. The changes from 0.5.3 to 0.5.4 are: - fixed 'group' setting according to Asterisk defaults. - use SetCallerPres(prohib_not_screened) instead of CallingPres(32) for CLIR. - full CallingPres support added. - use mutex when debug/verbose messages are printed. - set dnid on incoming call. - catch errors in wrong dialstring. - set correct DIALSTATUS and HANGUPCAUSE. - set PROGRESS and PROCEEDING when the network signals them. - increased voice send buffer a little bit. - fixed seg-fault when unallocated number was dialed. Have fun Armin -- Cytronics Melware Weinbergstrasse 39 55296 Loerzweiler / Germany Tel: +49 6138 98110-0 Fax: +49 6138 98110-9 mailto:[EMAIL PROTECTED] http://www.melware.de ___ 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] DTMF transparancy
Good Day list, Does anyone know if it is possible to setup asterisk such that it passes DTMF Tones through from One channel to the next transparently. I have a situation where asterisk is answering the phone on Channel 1 (first channel of a PRI) and then bridges this call to Channel 25 (first channel of T1 connecting in a channel bank). I need to have asterisk NOT do anything to the incoming DTMF from channel 1. this audio tonbe needs to be sent transparently through the bridge to the analog device on Channel 25. Any ideas. The problem I have currently is that DTMF tones are received from the remote unit calling into channel 1 (at 100ms per tone) however asterisk is detecting this tone and retansmitting it to channel 25 (at what appears to be 500ms per tone). I need the tones to be heard by the device connected to channel 25 EXACTLY as the remote unit is sending them. Thanks for your ideas, solutions or moral support. J ~ron oledata.mso Description: Binary data ___ 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] InfoWeek Article on VOIP
On Sat, 2005-07-16 at 10:12 -0700, Michael D Schelin wrote: I agree with you but not 100% with them. An IP to Ip call on an ATA flat out is better . Now don't even get me started about cellular. My Service dosen't drop calls in the middle of conversations. VoIP is a notch better than Cellular. What a lot of people dont consider with VoIP is the qualiuty of their ISP and how well connected their ISP is to everything else. My ISP for example (only game in town that isnt dialup) has 1 feed from sprint, I am guessing a T3 (I live in a rural area) and no QoS of any kind. So in general they suck for VoIP because of the latency they add to the link. Many people I have talked to think internet access is internet access and the contention rate is never thought of. This greatly affects any review of VoIP. Of course a private IP network (again a lot of people think VoIP as voice over the internet not thinking about private networks) is usually better because it can be tweaked for voice apps specifically. Even if you dont have a private network adjusting packet size and jitter buffers for that link specifically can increase performance. It ends up being more than just tossing a box on the net with asterisk or whatever on it. Now that I think about it I havent looked anywhere for network performance tuning for voice apps, does voip-info have a wiki page? If not perhaps it should with general properties based on link types and all and possibly specifics for certain operating systems and/or network equipment. Since updating wikis is against my religion I am unable to do this (strict religion, forbids me from contributing to any GPL project - forced to release my code BSD style if free, or updating wikis). But there are enough people that do not follow the same religion as me. That may help with performance all around, and increase the user experience. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Cisco 7960 Auto Answer (SIP)
1) Trying to create a browser based Click-to-Call feature for * that appears to the user as a hands free call on Cisco 7960 phones (SIP). If I use the Action: Originate function, the phone does not auto answer, but rather rings and if answered initiates the call. If I manaually change the line to auto answer (Intercom)in the 7960 configuration, it auto answers, but all calls to that extension auto answer. I know the work around with a dedicated extension for originating calls, but was hoping for a better solution. Currently, Covad appears to have an auto answer feature that is software controled, as it works on these Cisco phones with them, (they are using MCGP). So at least I know it is possible. I have found very little documentation using * with MCGP. Anyone have a solution? 2) Is there a commerial grade termination service for toll free numbers that is toll free? I am looking at 25 to 75 sustained (outbound) calls per hour. Most of the LD termination services appear to charge the same for LD as toll free numbers in the US. ___ 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] Re: Any Ideas??? 3rd time posting = Sipura SIP Phones Multi-Line
Steve Gladden wrote: Still looking for some direction with this subject: I think the term is called multi-line appearance Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to indicate someone is using 'line 1' and they should NOT be able to pick up 'line 1' so long as that 'line1' is in use by another phone. You are trying to emulate a key system, Asterisk is a PBX. I don't you will get this done as there is no concept of lines, all phones are extensions. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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] Voicepulse connect - unable to dial out, asterisk says 9696
Hi, for some weeks now I have been unable to make calls via my voicepulse connect IAX account? When I attempt the console looks like this:- rt*CLI -- Executing Dial(SIP/2008-cf55, IAX2/NBhXX:[EMAIL PROTECTED]/12124565900) in new stack -- Called NBhX:[EMAIL PROTECTED]/12124565900 -- Call accepted by 66.234.228.160 (format ulaw) -- Format for call is ulaw -- Hungup 'IAX2/66.234.228.160:4569/1' -- Executing HasNewVoicemail(SIP/2008-cf55, 2002) in new stack rt*CLI and Asterisk speaks back to me 96 96 And thats it!? I'm not aware I changed anything at this end. Asterisk 1.07. Thanks Mike ___ 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] Asterisk Interface with mobile phone
Hi: I live in a country where calls from landline phone to a mobile phonesis more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to thepstn and mobile phones. I want to save money by transferingmobile calls througha mobile phone. Is there some interface between the FXOcard and the mobile phone so asterisk can dial the mobile phone? Lnadlines and mobile phonescan be differntaited by theirprefix. Thanks Start your day with Yahoo! - make it your home page ___ 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 Interface with mobile phone
On Sat, 2005-07-16 at 11:55 -0700, chawki hammoud wrote: Hi: I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can dial the mobile phone? Lnadlines and mobile phones can be differntaited by their prefix. Thanks There are a few ways to do this.. chan_bluetooth is one if you have a mobile that speaks bluetooth. www.cellsocket.com if you have a compatible phone. And some others have previously mentioned (and I forgot, the archives have it or maybe the posters would be nice and repost) devices that work like cellsocket. Basically cellsocket and similar devices are a charging base station for mobiles that provide a FXS port on the mobile side. Connect it to an FXO port and you can use it as a normal phone. Additionally there are internal devices that work directly as a mobile but is fixed in the computer (pccard or usb typically) if you dont have the need to remove the mobile and take it with you when you arent at home/office. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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 Interface with mobile phone
Hello try to setup a gsm gateway it will do what you want best regards Thierry De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de chawki hammoudEnvoyé: samedi 16 juillet 2005 20:55À: Asterisk-Users@lists.digium.comObjet: [Asterisk-Users] Asterisk Interface with mobile phone Hi: I live in a country where calls from landline phone to a mobile phonesis more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to thepstn and mobile phones. I want to save money by transferingmobile calls througha mobile phone. Is there some interface between the FXOcard and the mobile phone so asterisk can dial the mobile phone? Lnadlines and mobile phonescan be differntaited by theirprefix. Thanks Start your day with Yahoo! - make it your home page ___ 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] Someone to have any idea how to run an Outbound Proxy?
Hi, Anyone to know how to run an Outbound Proxy to solve the NAT problem? I saw the FreeWorldDialup (FWD) are using a SER on port 5082. I have tried to configure SER with nathelper/rtpproxy. Anyway I still can nothave a callfromSIP UA behind a NAT but in same time it works perfect with the FWD's Outbound Proxy. Maybe I am not in right way - please help. If anyone know how the FWD's Outbound Proxy is configured it will be great. ___ 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: Any Ideas??? 3rd time posting = Sipura SIP Phones Multi-Line Appearance... How to use? |-----WAS---- [Asterisk-Users] NEWBIE Question: Asterisk with multiline/button phones
Generally speaking one works against one's own best interests when he reminds the group that he has been posting on a topic repeatedly without anyone answering. What you are asking for is not reasonable; it's not the way Asterisk works, and there is in my mind (and I'll bet in the minds of others) no logical reason to need what you're asking for. Perhaps the response is so that I can construct an Asterisk system that has 100% of the exact behavior of a key system. But if that's what you want, and that's all that you want, and you must have exactly that, get a key system. Asterisk is the wrong tool for the job. B. ___ 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] arrgg! www.voip-info.org down again (or too busy)
Lists skrev: On Friday 15 July 2005 16:54, Peter Osborne wrote: You can alway use google's cache. Use site:www.voip-info.org when searching or type the full URL into google and click on the cached version. Pete On 15 July 2005 4:36 pm, Damon Estep wrote: Does anyone have a mirror of this running? Yes, it's annoying as hell. A few times I've been close to make a mirror for myself so that I can access it reliably. Not quite the same to go through google. I just have to figure out what it would take set one up... I've read yours and others among with you and I totally agree with you. I've mailed the maintainer of the site last night, here's what he responded: Thanks for the kind words. There are no current mirrors -- wikis are somewhat difficult to mirror. I'm working on new software, which I hope will both improve performance, and make it easier to mirror. Jim James H. Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] - Original Message - *From:* Johan Nordström mailto:[EMAIL PROTECTED] *To:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Friday, July 15, 2005 2:40 PM *Subject:* Mirroring? Hi there, we are alot of people using voip-info.org and thinks it is perfect! The problem is that somethimes the site is unresponsive or just too busy. Are there any mirrors availible, otherwise I'd (with many more) would gladly help you with mirroring of the site. Best regards, Johan Nordström Sweden -- There are 10 kinds of people in this world: Those who can count in binary and those who cannot. So it seems that we're able to put up a mirror (or mirrors) in the not far future. Anyone interested? I'll keep in touch with the author/-s so I'll know when he/they are done with their new software! What I figured the requirements would be Twiki, PHP and a mySQL-database. Best regards, Johan ___ 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 registration question
Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they think packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that from a phone this works fine, and that our config must be at issue. The claim is that Asterisk isn't doing MD5 authentication right, and since I'm not an expert with SIP MD5 auth in asterisk, may be true. Right now, I'm trying to get the registration happening. On a test server, we've been able to put through a call w/o registration, so it seems some of this can be compatible. I'm wondering if I can use md5secret with a register = statement. The current busted config: [general] ;register = userid:pass:[EMAIL PROTECTED]:5069 [myipsolution] type=friend authuser=acctid username=userid secret=pass md5secret=XXXMD5HASH of userid:asterisk:pass X nat=yes host=voipprovider.com port=5069 insecure=very canreinvite=no The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' The password is right, as given and verified by the provider. Any suggestions would be great. Thanks, J. ___ 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] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
Try terminating using 50 ohm resistors as suggested by this guide: http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html in chapter 2.2 (Connect ISDN telephones to your ISDN card.) Best regards, Jan Snelders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? Carl Andersson wrote: Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? Hi Carl, I just started yesterday afternoon with exactly the same setup so you are a bit ahead of me. If anyone answers you directly then please be kind enough to forward their comments to me. I have not even tried to sort out trunks, bristuff or anything yet but it might be worth pointing out that my initial problems were that, using an old pIII motherboard with pci graphics, network plus 2 HFC bri isdn cards, I ran out of IRQs. I had to lock down exclude the irq for the network card before the 2 ISDN cards woke up. I now have the network 1 ISDN card on their own IRQ and the graphics and 2nd ISDN card sharing an IRQ. Maybe this could be a similar problem for you? I'm using this HW with AAH 1.1 Keep in touch, Cheers, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ 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
Re: [Asterisk-Users] Sip registration question
On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they think packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that from a phone this works fine, and that our config must be at issue. The claim is that Asterisk isn't doing MD5 authentication right, and since I'm not an expert with SIP MD5 auth in asterisk, may be true. Right now, I'm trying to get the registration happening. On a test server, we've been able to put through a call w/o registration, so it seems some of this can be compatible. I'm wondering if I can use md5secret with a register = statement. The current busted config: [general] ;register = userid:pass:[EMAIL PROTECTED]:5069 [myipsolution] type=friend authuser=acctid username=userid secret=pass md5secret=XXXMD5HASH of userid:asterisk:pass X nat=yes host=voipprovider.com port=5069 insecure=very canreinvite=no The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' The password is right, as given and verified by the provider. Any suggestions would be great. Hi, Did you try to put the md5 encoded password in your register= line ? -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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
Re: [Asterisk-Users] Someone to have any idea how to run an Outbound Proxy?
I think FWD is using Jasomi's SBC to tackle NAT issues. On 7/16/05, Shady [EMAIL PROTECTED] wrote: Hi, Anyone to know how to run an Outbound Proxy to solve the NAT problem? I saw the FreeWorldDialup (FWD) are using a SER on port 5082. I have tried to configure SER with nathelper/rtpproxy. Anyway I still can not have a call from SIP UA behind a NAT but in same time it works perfect with the FWD's Outbound Proxy. Maybe I am not in right way - please help. If anyone know how the FWD's Outbound Proxy is configured it will be great. ___ 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
Re: [Asterisk-Users] Sip registration question
Hi, Quoting Michiel van Baak [EMAIL PROTECTED]: On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' Did you try to put the md5 encoded password in your register= line ? I didn't before (I wasn't sure that was a valid syntax) ... but I have tried now, same error. Is there something to tell asterisk to try an MD5 auth, either in the password or on the registration line? Thanks for your quick response. J. ___ 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] PRI got event: HDLC Abort (6) on Primary, D-channel of span 1
Hey Kevin,
I managed to resolve this error after a week of pulling out my hair.
Here is what I did to resolve the error and a link below for further assistance.
1 - If you are not using 2.6 kernel, upgrade.
2 - Check your span line in your zaptel.conf. You should be receiving timing,
not giving it, when using a PRI (generally). Change the second number from 1
to 0. Save and restart asterisk. (span=1,0,0,esf,b8zs)
3 - I had a SATA RAID ARRAY setup (RAID 5) because I thought, hey, I can have redundency so that I don't loose voicemail if a drive crashes. Sadly if you run anything other than an IDE Drive you cannot use DMA and the Digium cards rely heavily on DMA. So I was forced to take out my RAID Controller and SATA drives and install a nice high end 120 Gig IDE Drive.
4 - In you BIOS disable any on board devices that you can (i.e. scsi
controllers, usb controllers, serial controllers, etc...). If your BIOS
supports/has an APIC(Advanced Processor Interrupt Controller) (which most
modern motherboards do) go into your IRQ settings and set them all to the
default AUTO type option. (you'll see why further below).
5 - In your 2.6 kernel enable these options:
PROCESSOR TYPE AND FEATURES ---
[*] Local APIC support on uniprocessors
[*] IO-APIC support on uniprocessors
DEVICE DRIVERS ---
ATA/ATAPI/MFM/RLL support ---
[*] Generic PCI bus-master DMA support
[*] Use PCI DMA by default when available
* VIA82CXXX chipset support
- I CHOSE THE VIA82CXXX CHIPSET SUPPORT FOR MY MOTHERBOARD, CHOSE THE
APPROPRIATE ONE FOR YOUR MOTHERBOARD, THIS IS IMPORTANT IN USING DMA. THE APIC
IS GOING TO ASSIGN IRQS AND PREVENT SHARING, ALSO FREES UP MORE THAN 16 IRQ
LIMIT. SAVE THE CHANGES AND RECOMPILE YOUR KERNEL.
6 - run these commands to enable dma if it is not already on and enable irq
unmask
linux# hdparm -d1 /dev/hda
/dev/hda:
setting using_dma to 1 (on)
using_dma= 1 (on)
linux# hdparm -u1 /dev/hda
/dev/hda:
setting unmaskirq to 1 (on)
unmaskirq= 1 (on)
linux# hdparm /dev/hda shows info.
/dev/hda:
multcount= 16 (on)
IO_support = 1 (32-bit)
unmaskirq= 1 (on)
using_dma= 1 (on)
keepsettings = 0 (off)
readonly = 0 (off)
readahead= 256 (on)
geometry = 65535/16/63, sectors = 80026361856, start = 0
linux# hdparm -i /dev/hda shows more info.
/dev/hda:
Model=WDC WD800JB-00FMA0, FwRev=13.03G13, SerialNo=WD-WMAJ97238449
Config={ HardSect NotMFM HdSw15uSec SpinMotCtl Fixed DTR5Mbs FmtGapReq }
RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=58
BuffType=unknown, BuffSize=8192kB, MaxMultSect=16, MultSect=16
CurCHS=4047/16/255, CurSects=16511760, LBA=yes, LBAsects=156301488
IORDY=on/off, tPIO={min:120,w/IORDY:120}, tDMA={min:120,rec:120}
PIO modes: pio0 pio3 pio4
DMA modes: mdma0 mdma1 mdma2
UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5
AdvancedPM=no WriteCache=enabled
Drive conforms to: device does not report version:
* signifies the current active mode
YOU CAN SEE ABOVE THAT UDMA5 MODE IS ACTIVE BY THE * NEXT TO IT.
7 - you can run hdparm -tT /dev/hda to get some benchmarks on your drive, mine
is running at:
/dev/hda:
Timing cached reads: 956 MB in 2.00 seconds = 477.59 MB/sec
Timing buffered disk reads: 126 MB in 3.03 seconds = 41.62 MB/sec
Which is literally 20 times faster than before I ran through these steps.
8 - Make absolutly positive that you are using the current most stable version
of asterisk.
9 - Lastly, if you are running any services that you can put on another machine
(i.e. TFTP, NTP) do so. Move them to another machine and try not to run any
services but what is absolutely necessary and asterisk.
This should take care of you but if you need more try the link below. When all
else fails, go to the digium supported hardware and change out your motherboard.
Fixing interrputs:
http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
Message: 6
Date: Fri, 15 Jul 2005 20:13:48 -0400
From: Kevin [EMAIL PROTECTED]
Subject: [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary
D-channel of span 1
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
I am getting an error in the log on the PRI span. The error is :
PRI got event: HDLC Abort (6) on Primary D-channel of span 1
I thought the problem was an interrupt conflict with the T110P card, so
I changed out the server to one that will dedicate the interrupt to the
T110P card. I still have the problem.
It's a dell 800 server with an SATA drive.
Can anyone offer
Re: [Asterisk-Users] Re: Any Ideas??? 3rd time posting = Sipura SIP Phones Multi-Line
Chris Mason (Lists) wrote: Steve Gladden wrote: Still looking for some direction with this subject: I think the term is called multi-line appearance Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to indicate someone is using 'line 1' and they should NOT be able to pick up 'line 1' so long as that 'line1' is in use by another phone. You are trying to emulate a key system, Asterisk is a PBX. I don't you will get this done as there is no concept of lines, all phones are extensions. Well, not exactly What he wants is reasonable and very common these days and is considered a hybrid PBX. Users want buttons and lights. They want key system operation with their PBX. Such systems have been around since telephone systems went from relays to electronics in the mid to late 70's The old concept of a PBX with extensions that were single line is long gone. Single line phones with dialing of codes is not easy to use. If it isn't easy to use there will be hell to pay. And in the PBX world an extension IS a line. Connections to the PSTN are called trunks. The terms line, station, and trunk are quite often mixed and misused . The 841 has a sharedoption in it's setup for each button , so it is probably is possible with SIP in some way. If Asterisk can't do it, then he will need to find another system that will, but it is certainly not an unreasonable request or feature to have if a system is to have a hope of success. John Novack ___ 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 registration question
On 17:01, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: Hi, Quoting Michiel van Baak [EMAIL PROTECTED]: On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote: The error on the console is: Jul 16 11:29:20 NOTICE[3361]:-- Registration for '[EMAIL PROTECTED]' timed out, trying again Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on authentication for REGISTER for 'userid' to 'voipprovider.com' Did you try to put the md5 encoded password in your register= line ? I didn't before (I wasn't sure that was a valid syntax) ... but I have tried now, same error. Is there something to tell asterisk to try an MD5 auth, either in the password or on the registration line? Thanks for your quick response. J. Hi, I don't think it is possible to use md5auth on register= lines. Have a look at: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf The one line that makes me think it is impossible is right below the Asterisk as a SIP client examples: Agreed, it's not very good to have a lot of cleartext passwords in this text file, but that's how it works now. If you find out I'm wrong, please send me or the list a reply -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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
[Asterisk-Users] VoIP with asterisk and x-lite
I have an OpenBSD 3.7 gateway. This gateway run Asterisk. I have two windows box which use X-Lite softphone, and each box connect to Asterisk using this softphone (X-Lite). Asterisk use the following configuration : /etc/asterisk/sip.conf ; Phone #1 [Phone1] type=friend host=dynamic nat=yes defaultip = 192.168.10.12 # windows box IP context = sip callerid=Phone1 1 ; Phone #2 [Phone2] type=friend host=dynamic nat=yes defaultip = 192.168.10.5 # second windows box IP context = sip callerid=Phone 2 i have the following extension : /etc/asterisk/extensions.conf [sip] exten = 1,1,Dial(SIP/Phone1,20,tr) exten = 2,1,Dial(SIP/Phone2,20,tr) One windows box have phone number 1 and the other windows box have phone number 2. I call Phone number 2 from Windows box with phone number 1 and work. The first session started (Windows box 1 call Windows Box2 ) , i talk on windows box 1 and i hear my voice on Windows box2. I close this phone session. I start a new session (Windows box1 call again Windows Box 2), i talk but now i don't hear my voice on windows box 2. I think this is a NAT issue . I don't hear my voice again until i restart Asterisk :( Any idea to help me ? Thank you very much ! ___ 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] InfoWeek Article on VOIP
On Sat, 16 Jul 2005 10:10:29 -0700, Bruce Ferrell wrote: Michael Graves wrote: Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves How do you see an ATA as different from and IP hardphone? As far as I can tell having the phone and ATA integrated isn't all THAT desirable, but that's me, I like to be able to choose the features on my phone and be able to connect it to the net... But that's just me. I have personally used Cisco ATAs, Sipura-2000s and 3000s. When I begin investigating switching to IP phones I tried Pingtel, Grandstream, Zultys, Snom and Polycom. To be fair I used each one for a couple of months, often as my primary desk phone if it looked like the device would cut it. I settled on Polycom 600s and 500s for my home office. I only have 5 phones. As someone who works from a home office professionally I feel that the call quality, multi-line capability and availability of serious business features are important. For a while , before I had a production * server, I had a pair of Sipura units connected to a 4 line Panasonic KSU system. The Polycom's simply sounded best, feel best in the hand, and have the on-board tools that I use daily. ATAs just don't go that far for me. I don't see it as having the phone and the ATA integrated. It's a SIP phone. Asterisk sees it as something slightly different than an ATA. I may have multiple registrations, of which several may be in use at once. It supports simple SMS stlye messaging. Heck the IP600 even has a micro-browser built into it, although I've not used this yet myself. I agree with others who have chimed in that IP-to-IP calls can sound better than PSTN calls. I have a co-worker who has a SipGate account in the UK. Calls to him via SipGate go out through my FreeWorldDialup account. They sound great. So good that in silent moments we often think that we've been severed, even with no silence suppression on the line. It really would be great to have a truly wideband codec available within Asterisk. I recall reading that the wideband version of iLBC is not released under GPL. Anyone know more about this? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] NAT Asterisk Peering
This is not a problem. I do this and a bit more. The IAX protocol helps quite a bit to go thru the NAT. -- Ted Serreyn Phone:262-432-0260 Fax:262-432-0232 Serreyn Network Services, LLChttp://www.serreyn.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Adams Sent: Thursday, July 14, 2005 10:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] NAT Asterisk Peering I was wondering and trying to figure out how the following idea would work and what might be needed to implement it, can anyone give any suggestions? What I am wanting to do is to setup an asterisk server, primarily so I can get voicemail, call files, and the idea of cheaper long distance by using the internet. What seems like the place that things could get tricky is that I am desiring to place the asterisk server behind a firewall that performs NAT operations on the packets. Any of the SIP phones or ATA devices that would connect directly to my box would also be behind the NAT location. The only passthru to the outside is in the idea of peering with other asterisk boxes, I do have one in mind, that are on the outside of my own NAT box. I am desiring to setup the peering with SIP rather than IAX at least on this main connection. Is there any examples of the VOIP-info website or other places that might be of use in this idea? Or does anyone have suggestions or knowledge of this working for them? -- Dan Adams - [EMAIL PROTECTED] http://www.infochi.net ___ 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
Re: [Asterisk-Users] InfoWeek Article on VOIP
On Sat, 2005-07-16 at 17:05 -0500, Michael Graves wrote: I agree with others who have chimed in that IP-to-IP calls can sound better than PSTN calls. I have a co-worker who has a SipGate account in the UK. Calls to him via SipGate go out through my FreeWorldDialup account. They sound great. So good that in silent moments we often think that we've been severed, even with no silence suppression on the line. One thing that many PSTN providers are doing for calls when they went digital is to insert small quantities of noise into the line. That way people do not think they are disconnected. There is a bunch of documentation on this, and even some that applies to VoIP providers/equipment doing the same (its basically a faint bit of white noise so you hear *something*). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] InfoWeek Article on VOIP
trixter http://www.0xdecafbad.com wrote: On Sat, 2005-07-16 at 17:05 -0500, Michael Graves wrote: I agree with others who have chimed in that IP-to-IP calls can sound better than PSTN calls. I have a co-worker who has a SipGate account in the UK. Calls to him via SipGate go out through my FreeWorldDialup account. They sound great. So good that in silent moments we often think that we've been severed, even with no silence suppression on the line. One thing that many PSTN providers are doing for calls when they went digital is to insert small quantities of noise into the line. That way people do not think they are disconnected. There is a bunch of documentation on this, and even some that applies to VoIP providers/equipment doing the same (its basically a faint bit of white noise so you hear *something*). It's sometimes called comfort noise... As far as I'm aware, it's only done in VoIP. I spent 15 years working with digital switches/T1 channel banks. I guess it might have been built in and I just didn't know about it, but we were very concerned about excess noise and quantization noise as it was. We used to inject a 1004 test tone and then use a notch filter to measure the amount of quantization noise at the reciever. Just as a by the by, G.711ulaw is the codec used in channel banks. ___ 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] InfoWeek Article on VOIP
On Sat, 2005-07-16 at 16:12 -0700, Bruce Ferrell wrote: It's sometimes called comfort noise... As far as I'm aware, it's only done in VoIP. I spent 15 years working with digital switches/T1 channel banks. I guess it might have been built in and I just didn't know about it, but we were very concerned about excess noise and quantization noise as it was. We used to inject a 1004 test tone and then use a notch filter to measure the amount of quantization noise at the reciever. I have seen it done at the switches that telcos use, not at the end user such as in a channel bank. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Implementing a ISDN home PBX
Hi, I would like to implement a inexpensive home PBX with Asterisk. I have an internal ISDN bus with 6 ISDN phones. I now thought, I connect a Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by my provider and a second Fritz card to the internal bus. Will this work? Thanks for the help, Arik ___ 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 Build from 16-7-2005 Crash! bug or what? ;-D
I have tried to update my CVS build from 29-6-2005 with a new one. but now when i start asterisk in verbose mode it crash after 1000+ lines of: ... ... Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-252000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-232000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-212000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-192000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-172000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-152000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-132000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-112000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-92000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-72000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-52000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-32000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257340.-12000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-992000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-972000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-952000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-932000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-912000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-892000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-872000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-852000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-832000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-812000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-792000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-772000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-752000 Jul 16 20:21:57 ERROR[23794] utils.c: warning negative timestamp -257339.-732000 ... ... i have rollback to the previous one (29-6-2005) and works perfectly. xad ___ 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] Cisco 7960 Auto Answer (SIP)
On 7/16/05, Asterisk Supporter [EMAIL PROTECTED] wrote: 1) Trying to create a browser based Click-to-Call feature for * that appears to the user as a hands free call on Cisco 7960 phones (SIP). If I use the Action: Originate function, the phone does not auto answer, but rather rings and if answered initiates the call. If I manaually change the line to auto answer (Intercom)in the 7960 configuration, it auto answers, but all calls to that extension auto answer. I know the work around with a dedicated extension for originating calls, but was hoping for a better solution. You can right a script that uses telnet to login to the cisco phone, and using the test command answers the phone. Currently, Covad appears to have an auto answer feature that is software controled, as it works on these Cisco phones with them, (they are using MCGP). So at least I know it is possible. I have found very little documentation using * with MCGP. Anyone have a solution? 2) Is there a commerial grade termination service for toll free numbers that is toll free? I am looking at 25 to 75 sustained (outbound) calls per hour. Most of the LD termination services appear to charge the same for LD as toll free numbers in the US. ___ 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
Re: [Asterisk-Users] Voicemail management
Just run somthing like this: rm -R /var/spool/asterisk/vm/default/1xx/* (I think this should do, otherwise something similiar will). On 7/16/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: For our hospitality system, voicemail management is an issue. I looked at vmail.cgi and it works for the user, but I need a higher level management capabikity, i.e., flush all email from extensions 1XX (Apartment1) when a guest checks out. Is there anything like that or does anyone want to work on this for me? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Voicemail macro?
This together with the other post doesn't make sense. Anyhow, such a macro will just do what the macro (err app) voicemail does. So why invent the airplane when it was done already. On 7/16/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: For our hotel application, we don't want to have to write 50 voicemail entries, is there a way to do a voicemail macro in the same way as a standard extension macro? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Voicemail management
C F wrote: Just run somthing like this: rm -R /var/spool/asterisk/vm/default/1xx/* (I think this should do, otherwise something similiar will). Yeah, I'm sittng around waiting for guests to check out! No, this is a job for php and an authenticated web page. Chris ___ 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] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? Hi Carl, I just started yesterday afternoon with exactly the same setup so you are a bit ahead of me. If anyone answers you directly then please be kind enough to forward their comments to me. I have not even tried to sort out trunks, bristuff or anything yet but it might be worth pointing out that my initial problems were that,using an old pIII motherboard with pci graphics, network plus 2 HFC bri isdn cards, I ran out of IRQs. I had to lock down exclude the irq for the network card before the 2 ISDN cards woke up. I now have the network 1 ISDN card on their own IRQ and the graphics and 2nd ISDN card sharing an IRQ. Maybe this could be a similar problem for you? I'm using this HW with AAH 1.1 Keep in touch, Cheers, Zoltan My problem turned out to be a termination problem. When using zaphfc together with other zap cards, it seems to be of importance in which order the drivers are loaded as well - At least in my case it would only work right if the X100P driver was loaded before the zaphfc driver. I have got verything working now, so if you have any questions you are more than welcome. You didn't write if you intended to use the ISDN cards to connect to ISDN lines, or if you wanted to create a setup like mine, with the card/cards in NT mode, acting as an ISDN switch of it's own. -- Greetings, Carl Andersson a.k.a Zaphod Beeblebrox Email: [EMAIL PROTECTED] - ICQ: 1705837. -=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=- ___ 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] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
Try terminating using 50 ohm resistors as suggested by this guide: http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html in chapter 2.2 (Connect ISDN telephones to your ISDN card.) Best regards, Jan Snelders I did something along the lines of that, and it works great now. But instead of terminating with 50 Ohm at one end of the line, I put 100 Ohm termination in both ends of the line... Thanks for the help! -- Greetings, Carl Andersson a.k.a Zaphod Beeblebrox Email: [EMAIL PROTECTED] - ICQ: 1705837. -=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=- ___ 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] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
On Sun, Jul 17, 2005 at 06:15:44AM +0200, Carl Andersson wrote: My problem turned out to be a termination problem. When using zaphfc together with other zap cards, it seems to be of importance in which order the drivers are loaded as well - At least in my case it would only work right if the X100P driver was loaded before the zaphfc driver. There seems to be some voodoo with zaphfc and ztcfg being run a number of times . Try disabling the post-install actions in /etc/modules.conf or /etc/modprobe.conf , and run ztcfg manually later. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Memory leak in asterisk CVS
Known issue. This was reverted later. Check the thread on the mailing list http://lists.digium.com/pipermail/asterisk-users/2005-July/116246.html Thanks, Erik On 7/16/05, Walter Klomp [EMAIL PROTECTED] wrote: Hi, My Asterisk CVS is apparently not doing much (other than keeping SIP IAX2 registrations alive and doing some ZAP calls (without echo-cancellation), but slowly the memory is filling up, so much so that 100m virtual memory is used up within 12 hours and I have to restart the asterisk application every 48 hours to make sure I have enough memory... How can I help resolve this problem? Problem occurs on both Sangoma and Digium installed systems. Fedora Core 3 and Centos 4.1 don't make a difference either. My version is Asterisk CVS-HEAD built on a i686 running Linux on 2005-07-11 16:29:02 I have removed the mailbox entries in my sip.conf which greatly reduced this problem. So, I suspect it may be in the sip or iax channel application. I also run quite a bit of agi scripts but none of them were alive when these memory-usage increases as shown below over a 1 minute interval with only 4 zap channels alive (2 calls) occured: ps -AF output... using this script: n=1;while [ 1 ]; do i=`ps -AF|grep ast|grep sbin|grep -v grep`; m=`echo $i|cut -f 6 -d\ `;if [ `echo $m` -ne `echo $n` ]; then echo $i; n=`echo $m`;fi;done root 15875 26881 0 15727 46240 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46248 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46256 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46268 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46280 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp root 15875 26881 0 15725 46288 0 10:01 ? 00:00:00 /usr/sbin/asterisk -fp Hope we can fix this somehow. Walter Klomp Singapore. ___ 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
[Asterisk-Users] beginners question about extension context
Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a given context Here are my contexts definitions: [from-sip] exten =101,1,Dial(SIP/phone1) exten =102,1,Dial(SIP/phone2) exten =103,1,Dial(SIP/phone3) [iax-user] exten=201,1,Dial(IAX2/phone4) exten=202,1,Dial(IAX2/phone5) If I try to call from IAX2 phone to say ext 102, I get request '[EMAIL PROTECTED]' does not exist I have tried to include iax-user in from-sip and I can make calls from SIP phones to IAX2 ones, but not the other way around. Now for an interesting bit. If I include from-sip in tthe iax-user, all is working fine -- I can make calls in any directions. If I try to do cross-include where one context is included into another and vise versa, IAX2 phone does not even register. Is there a better than include way to route calls between contexts? Thanks, Rudolf ___ 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
