[Asterisk-Users] Re: Zaptel compile error - unresolved symbols
Message: 12 Date: Fri, 17 Sep 2004 20:35:32 -0400 From: Rollo Tomnasi [EMAIL PROTECTED] Subject: [Asterisk-Users] Zaptel compile error - unresolved symbols To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hello - any help is greatly appreciated. I am trying to compile zaptel on debian 2.4.26-1-386. I have a single X100P card installed. When I run '/usr/src/zaptel/make clean;make install' I get the following: depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/zaptel.o /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/zaptel.o [ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf When i run depmod -ae: depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/torisa.o depmod: __write_lock_failed depmod: *** Unresolved symbols in /lib/modules/2.4.26-1-386/misc/zaptel.o depmod: __write_lock_failed depmod: __read_lock_failed Can anyone point me in the right direction? Thanks! make sure you have module versions enabled in your kernel and the correct .config darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100p on VIA EPIA-V
Date: Sun, 19 Sep 2004 12:59:52 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: [Asterisk-Dev] X100p on VIA EPIA-V problems To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 I'm running Asterisk on my (new) VIA EPIA-V motherboard. This seems to be the ideal platform for a home version of asterisk - its small, quiet, low power, and should have plenty of computing horsepower if only it would work! I'm running Redhat 9, Kernel 2.4.20-8, and Asterisk from CVS-HEAD-07/07/04-21:01:10 The phones in my house talk to asterisk via a Sipura SPA-2000. I have a X100p card (not from Digium, I regret, but one of the OEM cards sold by Diginetworks). Here' is a snipit from the boot log: I've been told very recently by a self-proclaimed linux expert (who happens to be involved with selling systems and motherboards, including the VIA) the VIA boards have a terrible PCI bus implementation that has caused lots of problems. The 'expert' has been involved with linux for years, is involved rather heavily in various audio apps, but has zero experience with asterisk. I don't have any experience at all with the VIA, so have no factual knowledge or experience. Simply passing on what I was told when I talked to him about a replacement motherboard. I wonder if I am seeing a similar issue, I am debugging a voice quality problem with a voiceronix openline4 on an VIA EPIA V mainboard. I get random tones chirps on calls through the FXO, otherwise it performs flawlessly. Anyone got any info on how to debug PCI issues? darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_vpb
Hi, Has anyone usingchan_vpb noticed that only one splash of ringback is provided to the PSTN? I have tried several different permutations in extensions.conf and interworking to mgcp sip and iax. I am using the Voicetronix supplied chan_vpb and asterisk from the 1.0 cvs source tree. thanks darren
RE: [Asterisk-Users] Asterisk Indications
From: Christopher Lee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk Indications Date: Sun, 25 Jan 2004 15:49:42 +1000 Organization: Data Chaos Reply-To: [EMAIL PROTECTED] Hi Steve, Interesting... I'm not sure! My copy of the original indications.conf had 400+17, and looking at the wiki it's the same there also http://www.voip-info.org/tiki-index.php?page=Asterisk%20indications%20defaul t I tested 400*17 and it made a difference, but I still think 420+400 sounds much closer to it... if there's any other Australian users who have customised the tones and want to try it out let me know what you think or what tones you're using. I have just been thinking perhaps the main advantage of letting the SIP device generate it's own indications is lower bandwidth use, for my setup this isn't really an issue, and if I can figure out how to directly modify the tones in the Cisco 7940 I'll have a go at it. You'll find the Australian tone specs here http://www.acif.org.au/ACIF/files/S002_2001.pdf cheers, darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Slightly OT and mildly insane: Modems through VoIP :-))
Message: 2 Date: Mon, 15 Dec 2003 23:13:46 +0100 From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Slightly OT and mildly insane: Modems through VoIP :-)) Reply-To: [EMAIL PROTECTED] Hi, First off, let me state that _YES, I am fully aware that what I am doing is insane, prone to major havoc and bad for general health_ :-)) Scenario: My GF needs an analog modem to use with her banking software (sodding backs don't supply a decent web-application for company use). I am experimenting to see if we can get it to work (albeit slow) trough our ATA186 talking g711 to Asterisk with chan_capi to the outside world. Should we fail, there are more sane alternatives, but humour me :-)) Now, with some modems on the other end I have received nice 26400bps handshakes, but it takes a long time and the successrate is about 25% :-) My feeling is this should be better if we choose to slow it down more. But who can tell me what the best modem settings would be to try ? My HAYES dialect is rather old :-)) Any experiences or hints are appreciated. -- Best regards, Florian Overkamp I've made data calls over g.711 voip from australia to the uk and they trained up to 31k2 fine. What will throw them is packet loss. Just one lost 20ms frame was normally enough to force a retrain. Lose a packet during retraining and the call would probably drop. Might pay to check the duplex settings on your ata and whatever its connected to. echo cancellers could possibly interfere with things but they should be disabled by the 2100hz tone at the front of the call. cheers, darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Howto to test asterisk applications - VoIP Testing Solution
Message: 1 From: Areski [EMAIL PROTECTED] To: Asterisk-Users Mailing-list [EMAIL PROTECTED] Organization: Date: 15 Dec 2003 13:06:18 +0100 Subject: [Asterisk-Users] Howto to test asterisk applications - VoIP Testing Solution Reply-To: [EMAIL PROTECTED] Hello All, Anybody can advice me some tools to test VoIP applications! Is there perhaps some applications that can generate a call every x minutes and analyze the call flow ? Thanks in advance, Areski You could use asterisk as a call generator by spooling outgoing calls from a cronjob. Or if you have budget, check out radcom performer or sunrise ghepardo. Both of these do a good job as a complete testing solution but they are not cheap. darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MS Messenger RTP
I have noticed some strange behaviour when using messenger as a sip client. Messenger appears to stop transmitting RTP like some sort of voice activity detection, and some applications on asterisk also respond by ceasing/not starting RTP transmission until they get something from messenger. Milliwatt and ringback tone are examples of applications that stop sending. Voicemail appears to keep transmitting regardless of what messenger is doing. Tried using different codecs but the problem was consistent. This is my first look at messenger as a sip client, so can anyone confirm what I am seeing here is normal? Using CVS from 12/08/03 and messenger 4.7.2009. Both asterisk and messenger are in the same subnet. cheers, darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk in a Centrex environment?
Message: 6
Date: Thu, 11 Dec 2003 08:18:53 -0500
From: Peter Pauly [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk in a Centrex environment?
Reply-To: [EMAIL PROTECTED]
Does anyone know what would be involved in making
Asterisk work as a voicemail system in a Centrex
environment? We have a Centrigram voicemail system
that belongs in the Smithsonian. There are analog
lines coming into the box and a 56KB data feed from
the phone company's switch.
Peter
Asterisk would need to be able to set the message waiting indicator. This is
often done over the SimpleMessageDeskInterface (that could be the 56k link)
I found this link in the archives
http://lists.digium.com/pipermail/asterisk-dev/2003-June/000884.html SMDI
has been mentioned as something good to have. SMDI has a lot of other
functionality, but for message waiting there are two directives required:
Set message waiting on :- OP:MWI {dn}! EOT
Set message waiting off:- RMV:MWI {dn}! EOT
These messages are usually sent via serial port or ascii tcp connection. The
method of message waiting (lamp, stutter tone etc) is implemented on the
switch.
An analog system would also need to understand the other parts of SMDI that
identify the forwarded party, I'm not too familiar with those. Better if you
could do PRI...
darren
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call does not terminate correctly
Message: 4 From: ProvoCityPower [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Mon, 8 Dec 2003 20:18:12 -0700 Subject: [Asterisk-Users] Re: Call does not terminate correctly Reply-To: [EMAIL PROTECTED] This a re-submittal hoping for some input: We are using an MGCP configuration. There seems to be some = incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is = how our client gateway Vendor sees it: 1. The first call is initiated. (CRCX) The interesting thing here = is that the CA (Call Agent) tells us to go directly into sendrecv mode = which means that we start streaming audio immediately. All other CAs = that we've worked with do not instruct us to go to sendrecv mode until = the number has been completely dialed. I agree * shouldn't really go to sendrecv until the B party has answered the call but I've assumed this is so treatment tones can be played (eg busy tone seems to be sent via RTP) 2. The call is terminated when hung up. The call agent responds to = this, but it never tells us to delete the connection and we continue to = stream audio. I don't see this behaviour in my setup. Does the call work on-net to another mgcp endpoint? This is how chan_mgcp ver 1.31 clears down a call to the asterisk milliwatt tone: endpoint asterisk = ntfy hd - -200ok -mdcx recvonly 200ok- -dlcx 250ok- You don't mention how you are accessing the PSTN? Are you interworking a couple of protocols here? 3. The next call is attempted. We are now, not in the state that the = call agent thinks we should be in and we are streaming audio to a UDP = port that is now closed since the CA tore down the first call. 4. The unit is rebooted. (The T2 is hard reset) The RSIP that is = sent to the call agent basically resets the state machine and now the T2 = and CA are in sync. =20 I'm not sure why this is happening, but maybe Asterisk can help. It's = clearly something in their code, but I can't really tell any more than = that. Our sequence of events: 1) Made first phone call to cell phone. Call was successful left it on = for a few minutes. Tried punching all kinds of digits while on the call. = Hung up. 2) Made second call. Picked up handset, was receiving dial tone. Tried = first digit and received the error (buzzing sound from the handset) . = The digit tone goes haywire and repeats itself over and over again (I = think this is what creates the buzzing tone). Tried to make call while = this was taking place. Hung up.=20 3) Reset T2. 4) Made three-four more additional calls all worked after resetting = T2.=20 Any input would be greatly appreciated. maybe a trace might help. darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP IADs
Message: 7 From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Organization: Telefonica CTC Chile Date: 06 Dec 2003 02:14:22 -0300 Subject: [Asterisk-Users] MGCP IADs Reply-To: [EMAIL PROTECTED] Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I´m trying to find an IAD which works with Asterisk. I´ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it´s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it´s nice to control using MGCP. Any information will be apreciated ! Thanks. -- Juanjo sin .sig Using Askey and they seem OK, not in a production environment though. darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IAD with MGCP
Message: 11 From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Organization: Telefonica CTC Chile Date: 03 Dec 2003 12:23:26 -0300 Subject: [Asterisk-Users] Cisco IAD with MGCP Reply-To: [EMAIL PROTECTED] snip hostname 192.168.65.200 [192.168.65.200] host = 192.168.65.200 I seem to recall a similar issue with a different IAD. Try changing the hostname and endpoint name to something else (like cisco2430) darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk behind NAT How to do it. (Leif Madsen)
I'm pretty sure that is incorrect. The inside_net is the ip address of the asterisk server, and the inside_mask is the subnet mask. At least that is how I have mine setup in my sip.conf, and it works. inside_mask for the internal mask would make more sense to me as well :) -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com In my configuration I have internal SIP clients registering from 192.168.0.0/28 and my * address is at 192.168.0.100. Using the host address of the * box as the inside_net variable the audio from 192.168.0.0/28 was sent to the outside_addr variable giving one-way speech. Setting internal_net to the subnet address of 192.168.0.0 and inside_mask to 255.255.255.0 the call behaved correctly. darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MGCP problem
Message: 1 From: Sergi Gabunia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 27 Nov 2003 12:05:15 +0400 Subject: [Asterisk-Users] MGCP problem Reply-To: [EMAIL PROTECTED] I have VOIP network built with MGCP endpoints.The manufacturer of = endpoints is ASKEY. I downloaded latest Asterisk software and found it = very useful for me. I configured it and it seems taht everything works = OK when I am testing it with one or two endpoints. After that I tried to = move Asterisk to working network and replace existing call manager. It = starts working and calls are proceeding but after a while I could not = hear a dialtone and saw in logs the following: Nov 27 11:40:57 WARNING[10251]: File chan_mgcp.c, Line 2127 = (handle_hd_hf): Unable to create switch thread: Interrupted system call I looked in chan_mgcp.c file and saw that this error occures after = pthread_create functions and it means that this system call was = interrupted permaturely with a signal before it was able to complete.=20 Please, help me to resolve this problem. Best regards, Sergi Gabunia I am using v1.29 of chan_mgcp.c with the askey unit and I can see a similar problem with memory not being released after off hook/on hook transition. Anybody fill me in on what debugging data would be useful in identifying this problem? darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk behind NAT How to do it.
Message: 9 From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Organization: http://www.hacklocalhost.com Date: 27 Nov 2003 23:10:42 -0500 Subject: [Asterisk-Users] Asterisk behind NAT How to do it. Reply-To: [EMAIL PROTECTED] Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.cStatus: Locally Modified Working revision:1.258 Repository revision: 1.258 /usr/cvsroot/asterisk/channels/chan_sip.c,v While in pwd /usr/src/asterisk/channels/ patch -p0 /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com I can confirm this works for my NAT'd setup as well. Just one comment though that the inside_net variable is your internal subnet address not the asterisk server address. cheers, darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Help needed with IAX behind NAT
snip In this configuration I had to forward the iax port on both NAT boxes to the * box. 192.168.0.100 -NAT1- publicIP - publicIP - NAT2 192.168.0.100 It wasn't sufficient to register from one * box to the other to get audio working because the source port was different. * was version 0.4.0 and the NAT box was netbsd/ipfilter 3.4.27 in this case. If things haven't changed too much, might be your problem, can your firewall admin forward iax ports to your gnophone? cheers, darrenmc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO/FXS hotline
I was thinking of a hotline set up something like this: FXO --*--IAX--*--FXS The dialtone has to be provided by the remote end and flash hook has to be transparent Anyone have experience with hotlines on *? Would this work? cheers, darren --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.509 / Virus Database: 306 - Release Date: 8/12/03 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP behind NAT
From: Humberto Atristain [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MGCP behind NAT Date: Wed, 30 Jul 2003 19:07:40 -0500 Reply-To: [EMAIL PROTECTED] My trouble is that the MGCP devices lost the connection with the asterisk My gateways are ASKEY MGCP Any comments? Humberto If the gateways are losing the connection intermittently, could the NAT be timing out the bindings? Seen a problem where you could not ring the gateway but could get dialtone and firewall was the problem. Someone correct me if I'm wrong, but * does not use AUEP to check gateway status regularly to keep the bindings active darren --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.504 / Virus Database: 302 - Release Date: 7/24/03 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
