Re: [asterisk-users] [SOLVED] checking 'voicemail externally - doesn't work
I do use just one context for voicemail. This is like the T-Mobile voicemail center (IE: 305-776-MAIL).voicemailservice.gsm is You've reached the voicemail system. If you have a voicemailbox on this system, press star. To leave a message for another customer enter their ten digit number now however customer should be subscriber.
[voicemail-service]exten = 4193016245,1,GoTo(voicemailaccess,1)exten = voicemailaccess,1,MailboxExists(${CALLERID(num)})exten = voicemailaccess,102,VoiceMailMain(${CALLERID(num)})exten = voicemailaccess,2,Background(voicemailservice)
exten = voicemailaccess,3,Background(silence/10)exten = voicemailaccess,4,GoTo(voicemailaccess,1)exten = *,1,VoiceMailMainexten = *,2,Hangupexten = _1NXXNXX,1,VoiceMail(u${EXTEN:1})
exten = _NXXNXX,1,VoiceMail(u${EXTEN})On 10/23/06, Joseph [EMAIL PROTECTED] wrote:
On Mon, 2006-10-23 at 17:59 -0400, Andrew Joakimsen wrote: On 10/23/06, Joseph
[EMAIL PROTECTED] wrote: I just try with single authentication DISA, doesn't work, password is not recognized. Try without any disa whatsoever
I think DISA has to be there as it gives access to internal dial tone,isn't it?I can be without password,[snip] Did you try exten = 1000,1,VoicemailMain() as I said above with NOTHING BETWEEN THE PARENTHASIS???
Thank you, Yes It Works! It works without parenthesis.I was trying to make make it to work with one voicemail context but inthis case I will create another voicemail_outside context without
anything between parenthesis for outside access.exten = 1000,1,VoicemailMain() In this case all internal callers can access their voicemailbox without password but when a call comes from an external source PSTN
line it is asking for password and it goes through correctly: vm_execmain: Specified user 'pstn1270' not found (check voicemail.conf and/or realtime config).Falling back to authentication mode.
(as the user pstn1270 is not in voicemail.conf file) but without the |s somehow it is distorting the caller ID from pstn1270 to
'tn127011' that is why it doesn't work, but I can not pin-point what is changing caller ID. You said the mailbox number is11 and the caller ID Is correctly
pstn1270 and incorrectly tn127011 since the mailbox number is 11, I don't see how fixing (what does your CDR say??) this issue will fix your voicemail issue. Why do you insist on using the caller
ID? Remember what you are trying to do, if user has to dial into the system from an outside phone their CALLER ID WILL NOT BE THEIR MAILBOX NUMBER.As I've mentioned above I was trying to get by with one [voicemail]
context but I guess I'll have two. For the last time, try: exten = 1000,1,VoicemailMain() inside your disa-access context, and get rid of the old voicemail
include statement. That will work, here is a detailed sequence of events Enter disa password, press # At the dial tone dial 1000 System says Comedian Mail. Mailbox?
You dail the mailbox number which you stated above is 11 So press the 1 key on your telephone, if you wish you can dial # after, if not just wait.
System says Password? You dial the password, if you want you can press # after it, if not just wait I'm not going to respond to this thread any more. I've given you step
by step EXACTLY what to do, anyone else would have gotten a USD 100 ++ bill for that advice.Thanks Andrew for your patience.--#Joseph___
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Re: [asterisk-users] Asterisk conferencing features
something like this in da dialplanexten = 0078,1,Answer()exten = 0078,2,Wait(2)exten = 0078,3,MeetMe(0078,idpMs)exten = 0078,4,Hangup()notes:- change 0078 to your incoming no, - so when you want to do the conference, just dial the defined extension number- or you can do a blind tranfer to the room (i.e invite)also conferencing feature is also doable on the phone, check out phones from SNOM and xlite On 10/24/06, Rafael Marangoni [EMAIL PROTECTED] wrote: Does anyone knows a simple how-to, to make sip conferencing on asterisk?2006/10/23, Rosli Sukri [EMAIL PROTECTED]: On 10/24/06, Rafael Marangoni [EMAIL PROTECTED] wrote: Hello! I'm new in Asterisk and I hope that my trouble is very simple. We're implementing a Education Project of a e-Learning system (LMS) that uses conferencing (video and audio) over internet. The e-Learning system will be on GPL license, and for that, we're using only free software to implement. Asterisk is our first choice for video and audio conferencing, and making tests, started to implement it. The questions are: 1. Asterisk makes sip conferencing? (I know the aswer is yes) yes, via the 'meet-me' application 2. Asterisk need Digium hardware to do that ? On asterisk handbook I found: Note that for technical reasons, you must have at least one Zaptel interface (of any kind) installed in your Asterisk system if you wish to use conferencing. (page 7) it needs it for 'timing'. on freebsd i have manage to install it without a physical zaptel card, by just loading the module to provide the timing 3. Asterisk make video conferencing? not yet.. it only supports video call i.e 2 party where as conference usually means more than 2 4. If yes, anyone have docs more detailed on how to do that? 5. Anyone know clients (softphones) under gpl that we can use the code to implement on this aplication? ekiga provides both audio and video capabilities, it is part of gnome. for windows you can use xlite its gratis software but not gpl I need asterisk only for internet conferencing, and I know that it's much more than that. Thanks, and sorry for the questions Rafael Marangoni ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 faxing with spandsp and Grandstream HT.486
Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator inititate a re-invite ? txfax as originator: T.38 fax exchange takes place but the transmission is not successful, txfax reports errorcode 60 (Disconnected after permitted retry). Can someone recommend a T.38 able ATA which is working with spandsp ? Are there any terminals known which has been tested against spandsp ? Thanks ! Best regards Hans P.S.: Asterisk 1.2.7.1 with patched T.38 patch and patched app_rxfax.c app_txfax.c and udptl.c. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Audiocodes MP-20x
I have used AudioCodes MP 102, 104 and 108, both FXS and FXO. I have also used AudioCodes Mediant 2000. I can tell you that these are good devices. There are also many other media gateways that have a lot of facilities, but many of these implement those facilities in software. AudioCodes has also a quite good lets say -- hardware support. I havent used MP20x. -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone:0744137020 email: [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Monday, October 23, 2006 1:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Audiocodes MP-20x Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UA - number assignment
My problem is simple and Ive issued it about 3 weeks
ago. I want the UAs to authenticate with a number to the SIP server. Is this
possible?
For example, I configured an AT-RG613TX (Allied Telesyn
Residential Gateway). In its configuration it is not possible for me to skip
specifying a number (ex. 102) along with the username. Ive looked into
the source code (SIP implementation) of Asterisk and, as I figured out, it is
not possible to tell Asterisk the number the user has.
The question is: how can I assign a number to a user in
Asterisk? One solution would be to define two rules in extensions.conf :
exten = 102,1,SetCallerId,${FWDCIDNAME}
exten = 102,2,Dial(SIP/pianas)
these would tell Asterisk that user pianas has the number
102.
Is there any other solution for my problem? (a database for
example).
Thank you.
--
Paul Ianas
Programming Engineer
Level 7 Software
Timisoara, 59D
Bucovinei
phone:0744137020
email: [EMAIL PROTECTED]
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[asterisk-users] mgcp registration with asterisk
HIi am trying to register mgcp gateways(Polycom 501, 601) to asterisk as a call agent, mgcp gateways are not registering to the call agent.Please help me on this if any one knows how to congigure the mgcp.conf on asterisk as well as an MGs.The following are the details of mgcp.conf on asterisk.mgcp.conf[general]port = 2427bindaddr = 0.0.0.0[0004f205c258] //MG MAC Addresshost = 172.21.67.137 //MG IP Address(static)context = defaultcanreinvite = noline = aaln/1Please Let me know if any one already tryied MGs registration with asterisk.Kind Regards,- Ashok P. Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk and HMP
Hi all, Does Asterisk now support Intel's HMP platforms ? Does it support in 1.4 version ? There's a special driver for Intel-based HMP hardware+software for ABE. On the other hand, Asterisk has always been doing HMP :). In fact, I would say Asterisk's success in HMP is one of the push factors for companies like Intel, NMS to move to HMP from their traditional DSP-based designs. Yes, i known that. Thanks for your reply, you confirm me that it is available only in ABE. Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Junghanns quadBRI and mISDN
Hi. I'm trying to run a Junghanns quadBRI card with mISDN drivers. I'm able to compile kernel mode user mode mISDN components as well as chan_misdn. The misdn-init config properly detects the card and starts the hfcmulti driver; lsmod shows all required drivers are loaded. However, the misdnportinfo seems not able to find any card. Has any one successfully managed to run Junghanns cards with mISDN? (there are a couple of serious issues using bristuff and we've been looking for alternate drivers). Thanks, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] newbie astdb error, please help
I am getting this warning:- Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value ' 192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23' in family 'SIP/Registry I checked the file permissions. They are proper. There doesnot seem to be a visible error. No change has been done in any conf files for the past 4 months. The reinvite has also stopped. I dont have any idea whats happening. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Core dumps when Releasing clone lock
Hi, Our Asterisk server has started core dumping more often, which seems to be related to when it releases the clone lock, and it stops masquerading the channel. So, I wondered if anyone had an idea what causes these masquerading attempts. Our system is running: Asterisk SVN-branch-1.2-r31555M. Full-log: Oct 20 11:09:37 DEBUG[57839] chan_sip.c: Assigning Replace-Call-ID Info [EMAIL PROTECTED] to REPLACE_CALL_ID Oct 20 11:09:37 DEBUG[57839] chan_sip.c: 202 Accepted (supervised) Oct 20 11:09:37 VERBOSE[57839] logger.c: -- Stopped music on hold on IAX2/provider-15 Oct 20 11:09:37 DEBUG[57839] channel.c: Scheduling timer at 0 sample intervals Oct 20 11:09:37 DEBUG[57839] channel.c: Planning to masquerade channel IAX2/provider-15 into the structure of SIP/XX-158e Oct 20 11:09:37 DEBUG[57839] channel.c: Done planning to masquerade channel IAX2/provider-15 into the structure of SIP/XX-158e Oct 20 11:09:37 DEBUG[57839] channel.c: Got clone lock for masquerade on 'IAX2/provider-15' at 0x94334cc Oct 20 11:09:37 DEBUG[57839] chan_sip.c: update_call_counter(XX) - decrement call limit counter Oct 20 11:09:37 DEBUG[57839] channel.c: Putting channel IAX2/provider-15 in 256/256 formats Oct 20 11:09:37 DEBUG[57839] channel.c: Released clone lock on 'SIP/XX-158eZOMBIE' Oct 20 11:09:37 DEBUG[57839] channel.c: Done Masquerading IAX2/provider-15 (6) Oct 20 11:09:37 DEBUG[57839] channel.c: Didn't get a frame from channel: SIP/XX-158eZOMBIE Oct 20 11:09:37 DEBUG[57839] channel.c: Bridge stops bridging channels SIP/XX-c3c3 and SIP/XX-158eZOMBIE Oct 20 11:09:37 DEBUG[57839] app_dial.c: Exiting with DIALSTATUS=ANSWER. Asterisk exited on signal 11. GDB: #0 0x282e1373 in pthread_testcancel () from /usr/lib/libpthread.so.1 #1 0x282d992e in pthread_mutexattr_init () from /usr/lib/libpthread.so.1 #2 0x28145450 in ?? () Regards Øyvind Albrigtsen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail maintenance
Jordan Novak wrote: Has anyone created a GUI for this. I am not sure what youre looking for but we developed a Voicemail Manager: = http://sip-syndication.com best regards, Arnd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Junghanns quadBRI and mISDN
Maybe I found the cause... My Junghanns quadBRI PCI subsystem ID is 0xB552 (that is, quadBRI version 2.0), while mISDN expects 0xB550 (quadBRI version 1.0) I'm wondering what differences lie in the two boards from a driver's perspective... I'll try to recompile mISDN by adding also subsys=0xB552 to the list of supported pci devices. I'm not very familiar with kernel drivers, so... good luck to me. Alberto. Alberto Pastore ha scritto: Hi. I'm trying to run a Junghanns quadBRI card with mISDN drivers. I'm able to compile kernel mode user mode mISDN components as well as chan_misdn. The misdn-init config properly detects the card and starts the hfcmulti driver; lsmod shows all required drivers are loaded. However, the misdnportinfo seems not able to find any card. Has any one successfully managed to run Junghanns cards with mISDN? (there are a couple of serious issues using bristuff and we've been looking for alternate drivers). Thanks, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail maintenance
Arnd Vehling wrote: Jordan Novak wrote: Has anyone created a GUI for this. I am not sure what youre looking for but we developed a Voicemail Manager: = http://sip-syndication.com best regards, Arnd Hello Vehling, This product of yours, does it manipulate, files on the Asterisk server itself? If yes, does that mean, this has to be installed on the same server as Asterisk? As for you, Jordan, you can very easily create GUIs for voicemail management, if you store your voicemails in sql db. www.voip-info.org/wiki/view/*Asterisk*+Voicemail+*ODBC*+*storage . cheerz - Ben. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE:Asterisk and dialer Running on Thin Clients
You can, but it will demand a lot of work. We now work above introduction of such decision on thin clients under control of thinstation. As софтофона it is used mozphone (front-end), from the thin client network_client (back-end). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE:Asterisk and dialer Running on Thin Clients
Sorry You can, but it will demand a lot of work. We now work above introduction of such decision on thin clients under control of thinstation. As softphone it is used mozphone (front-end), from the thin client network_client (back-end). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] INVAL Messages
Hi Marian, Thanks for the info, something I'll check into... we have recently swapped the router over.. We receive a NEW message: THEM US(new) So the port forward (inbound) works ok.. We send them a reply: THEM US(AUTHREQ) And then they send us the INVAL... THEM US(INVAL) Now if any of the SRC ports got changed incorrectly, then wouldn't the call stop completely? This only happens some of the time, intermittently - is yours constant? I've a Draytek 2910G router.. Adrian Marsh Tel: +44 (0) 20 71833427 Fax: +44 (0) 1793 441594 http://www.ubiquisys.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marian Rychtecky Sent: 23 October 2006 20:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] INVAL Messages Hi Adrian, are you using this IAX thru NAT? I have this problem when i try call with IAX2 and this Asterisk server is behind the NAT... I think its here problem with UDP source port which is changed in NAT router, but im not sure 100% Marian Adrian Marsh napsal(a): All, Has anyone seen INVAL messages on an IAX link before? I'm occasionally getting them from my Gateway provider, and I need to narrow down the potential cause. Symptoms are: Incoming calls fail, I see NEW, AUTHREQ then INVAL messages between the two A*k boxes... then for no reason at all it'll start working ok again.. My Asterisik: 1.2.10, Gateway A*k : 1.2.0- Any known issues with IAX on either? My best guess so far is that the packets are getting corrupted on-route.. and I've asked the gateway folks to capture the traffic when it happens again to confirm... Thanks, Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marian Rychtecky [EMAIL PROTECTED] Tel. +420 724 397 441 ICQ 76582857 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] INVAL Messages
Hi Adrian, yes this problem has happen only sometime, but i dont know exactly when i has discover this - plase read my comments: A)Calling directly via public IP's (port 4569 is forwarded on ADSL modem to asterisk1) - not working Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 4 DCall: 0 [213.160.177.186:4569] - here is source port of transmiting packet 4569 (my site) VERSION : 2 CALLED NUMBER : 1299 CODEC_PREFS : () CALLING NUMBER : 1199 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Marian_Rychtecky LANGUAGE: en USERNAME: some_username FORMAT : 2 CAPABILITY : 2097151 ADSICPE : 2 DATE TIME : 2006-10-18 10:16:14 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 6ms SCall: 3 DCall: 4 [213.160.177.186:9785] -- but here i got response from port 9785 (other site, because of NAT translation on other site) AUTHMETHODS : 3 CHALLENGE : 585590037 USERNAME: VALSABBIA-SLOVENSKO Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 3 [213.160.177.186:9785] B) calling thru openvpn - working Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 4ms SCall: 1 DCall: 0 [192.168.255.2:4569] -- here im sending the same packet to port 4569... VERSION : 2 CALLED NUMBER : 1299 CODEC_PREFS : () CALLING NUMBER : 1199 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Marian_Rychtecky LANGUAGE: en USERNAME: user_name FORMAT : 2 CAPABILITY : 2097151 ADSICPE : 2 DATE TIME : 2006-10-18 10:14:16 -- Called VALSABBIA-SLOVENSKO:[EMAIL PROTECTED]/1299 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00012ms SCall: 1 DCall: 1 [192.168.255.2:4569] --- here i got reply from port 4569 (and this port is excepted) AUTHMETHODS : 3 CHALLENGE : 186694617 USERNAME: VALSABBIA-SLOVENSKO Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00034ms SCall: 1 DCall: 1 [192.168.255.2:4569] MD5 RESULT : b0674601456416db7e474de9a858c742 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT so get ACCEPT and evething is working Timestamp: 00041ms SCall: 1 DCall: 1 [192.168.255.2:4569] FORMAT : 2 So i suppose that is the problem with comparing of source and destination port of IAX2 packets Look at your tcpdump and write me plase if you have the same troubles. Good luck, Marian Adrian Marsh napsal(a): Hi Marian, Thanks for the info, something I'll check into... we have recently swapped the router over.. We receive a NEW message: THEM US(new) So the port forward (inbound) works ok.. We send them a reply: THEM US(AUTHREQ) And then they send us the INVAL... THEM US(INVAL) Now if any of the SRC ports got changed incorrectly, then wouldn't the call stop completely? This only happens some of the time, intermittently - is yours constant? I've a Draytek 2910G router.. Adrian Marsh Tel: +44 (0) 20 71833427 Fax: +44 (0) 1793 441594 http://www.ubiquisys.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marian Rychtecky Sent: 23 October 2006 20:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] INVAL Messages Hi Adrian, are you using this IAX thru NAT? I have this problem when i try call with IAX2 and this Asterisk server is behind the NAT... I think its here problem with UDP source port which is changed in NAT router, but im not sure 100% Marian Adrian Marsh napsal(a): All, Has anyone seen INVAL messages on an IAX link before? I'm occasionally getting them from my Gateway provider, and I need to narrow down the potential cause. Symptoms are: Incoming calls fail, I see NEW, AUTHREQ then INVAL messages between the two A*k boxes... then for no reason at all it'll start working ok again.. My Asterisik: 1.2.10, Gateway A*k : 1.2.0- Any known issues with IAX on either? My best guess so far is that the packets are getting corrupted on-route.. and I've asked the gateway folks to capture the traffic when it happens again to confirm... Thanks, Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list
Re: [asterisk-users] Unique call ID's across several systems
Matt Florell wrote: On 10/24/06, Steve Edwards [EMAIL PROTECTED] wrote: On Sat, 21 Oct 2006, Jeremy McNamara wrote: Steve Edwards wrote: I have a farm of 7 1u's) with te410p's. When a call comes in, I call an AGI that creates a channel variable named GLOBALID. The GLOBALID is 2 digits for the host number, 2 digits for the channel number, and 8 characters for the UNIQUEID encoded in hex. Then, I stuff it into CALLERID(name) so it will be available as the call is sent (using dial()) to the application servers. AGI is not going to scale. Why not do it all with dial plan logic? Store a global system id for each system and concat it onto the uniqueid CDR value. Our need for a globalid (in addition to our needs) was mandated by our credit card processor. They also limited us to length, upper case letters and digits -- not even a lowly period was acceptable to them :) To fit the host, channel, and unique id in I decided to encode the unique as hex. This system was designed before dialplan functions and I didn't see an easy dialplan way to encode to hex, so I cranked out an AGI. I don't know how fast a dialplan implementation would be (if a hex function was available), but I benched the AGI approach and the server does 1,000 in about 4 seconds. I'm sure your knowledge of telephony far exceeds mine, Jeremy. For a system with only 92 channels can the telco deliver more than 92 calls in 0.4 seconds over PRI? The average call duration for this system is about 10 minutes, so the probability of all 92 channels being free at the same time is rather small. I always wondered how much overhead invoking an AGI incurred. Setting up the AGI environment, spawning a new process, parsing the AGI environment, doing something useful, terminating the AGI process -- it sounds like it would take forever. Now I know it only takes about 40 msec. Have you tried using a FastAGI server script instead of a stand-alone AGI script? We saw dramatic improvement in speed and performance when we switched our logging functions to FastAGI from AGI. MATT--- We did as well. The performance of a local standard AGI decreased over time. I didn't have the time to figure out why so we just switched to fastagi on a remote host (Windows Service) and applied the patch that allows for n + 101 priority jumping in case of fastagi failure. HUGE improvement. Since then, I have had the opportunity to get root on a commercial asterisk black box solution and they also use fastagi but it is running on the localhost (127.0.0.1). I have not looked at the agi yet but assume that running it locally is even smarter. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF that lists UK ring etc tone settings, port impedances, disconnect tone settings and so on, but I'm still not getting PSTN hangup detection to work. Any help would be appreciated. Thanks, Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and HMP
Leo Ann Boon wrote: Gregory Duchatelet wrote: Hi all, Does Asterisk now support Intel’s HMP platforms ? Does it support in 1.4 version ? There's a special driver for Intel-based HMP hardware+software for ABE. On the other hand, Asterisk has always been doing HMP :). In fact, I would say Asterisk's success in HMP is one of the push factors for companies like Intel, NMS to move to HMP from their traditional DSP-based designs. I think this is now the Eicon HMP platform. It looks like Eicon bought this when the fools paid good money for Dialogic. Its amazing how many companies have got on the HMP bandwagon since we started the Zapata work in 1999. If you do a Google search you can find something like 10 companies promoting HMP type products. Few look like coherent products, though. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR_DISPOSITION_FAILED - Call has been answered correctly
Hi guys, I've an asterisk 1.2.5 runing as production system. Now it becomes very important to my customer an exact analysis of CDRs for their QoS to their customers. I've been analysing the CDRs, and i notice many entries like this: Calldate |Channel|Source | Clid | Dst | Disposition | Duration --- 2006-10-24 10:10:24 | Zap/2-1... | 2023| MSN: 2023| 100 | FAILED | 01:41 --- There are no complainings about dropped calls or something else. I must say, this is a ringgroup call, and this took me into this bug: http://bugs.digium.com/file_download.php?file_id=9084type=bug Thanks to Mark Spencer for the attached file that appears to fix this bug which apparently was only happening when the Dial statement contained more than one SIP user and when those SIP users were not connected. Is there someone else out there using this patch on production system? Problem Solved? -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote: Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF that lists UK ring etc tone settings, port impedances, disconnect tone settings and so on, but I'm still not getting PSTN hangup detection to work. I got an spa-3000 that works perfectly well now. (UK) I had some trouble at first though. What firmware are you using and what's the symptom? does it not hang up or does it hang up during calls? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax problem
On Mon, Oct 23, 2006 at 08:23:12PM -0700, Lee Howard wrote: If you don't mind saying, what is missing for full t.38 support? Steve giving Digium a royalty-free license to his GPL software or a pure-GPL branch of the Asterisk codebase, take your pick. Why royalty-free? AFAICS there's nothing to stop Digium licensing this code commercially from him, if it adds value to the product. Regards, Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UA - number assignment
On Tue, Oct 24, 2006 at 10:22:33AM +0300, Paul Ianas wrote:
My problem is simple and I've issued it about 3 weeks ago. I want the
UAs to authenticate with a number to the SIP server. Is this possible?
For example, I configured an AT-RG613TX (Allied Telesyn Residential
Gateway). In its configuration it is not possible for me to skip
specifying a number (ex. 102) along with the username. I've looked
into the source code (SIP implementation) of Asterisk and, as I
figured out, it is not possible to tell Asterisk the number the user
has.
The question is: how can I assign a number to a user in Asterisk? One
solution would be to define two rules in extensions.conf :
exten = 102,1,SetCallerId,${FWDCIDNAME}
exten = 102,2,Dial(SIP/pianas)
these would tell Asterisk that user pianas has the number 102.
Is there any other solution for my problem? (a database for example).
I'm probably misunderstanding the problem. Firstly, you can always use a
number as the SIP username if you like:
sip.conf
[102]
... parameters for phone 102
extensions.conf
exten = 102,1,Dial(SIP/102)
But this is generally frowned upon, because it's harder to manage in the
long term, particularly when people move offices, or you need to change your
numbering plan. Many people recommend using the MAC address of the phone as
its SIP username, as that is unique and stays with the phone forever.
Secondly, when you say you must specify a number along with the username,
you'll have to check how this maps to actual SIP parameters. There is no
agreed terminology for this, and many ATAs are really confusing in this
regard.
For example, I have a Speedtouch 716g router/ATA, and its VoIP parameters
are displayed as:
Phone Number Caller Name User Name Password
Line 1 __ ___
Line 2 __ ___
By experimentation, I determined that by Phone Number it means SIP ID,
the username part of the phone's SIP URI. Caller Name is the comment
string, and User Name is the authentication username.
So if I configured it as:
Line 1 foo bar baz bap
Then I would get:
From: bar sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
and it would authenticate as username 'baz', password 'bap'.
Most SIP implementations assume the SIP ID and the auth username are the
same, so I have to put the same in columns 1 and 3. (This is true for
Asterisk by default; I expect you can change the auth username for a peer
but I don't know how)
Maybe you're just suffering a similar terminology problem with this ATA.
So, if the SIP channel is defined as [pianas] then I'd try entering 'pianas'
into both the phone number and user name fields. If there is a Caller
Name or Caller ID field then you can enter whatever you like there; you
could enter 102, or enter Fred Bloggs. Many SIP phones will display the
value given by the far end when an incoming call arrives.
HTH,
Brian.
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[asterisk-users] Becoming a User on IRC
Hello, I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? Thanks, Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] something about Agent Transfer
Hi, all, we have some deployed Asterisk PABX, and we provide our customers some customized queue report, they report a problem when agent transfer call, the call duration includes the call time between the transferer and transferee. They use cisco 7940 phone and use the phone attended tranfer feature. We tested, it is really like that, and if they use asterisk builtin transfer feature, the queue log can have a explicit Transfer Event, but it is a difficult to convince the user to use asterisk builtin feature, that is not that intuitive like the buttons shown on the phone. Do you guys encouter the same problem? Any work around? or is the asterisk 1.4 solve this problem? -- Regards! Liangliang ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Conrad Wood wrote: On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote: Do any *UK* users have an SPA3102 (the newer version of the SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call has hung up? I've read everything I can find, including an SPA3000 UK setup PDF that lists UK ring etc tone settings, port impedances, disconnect tone settings and so on, but I'm still not getting PSTN hangup detection to work. I got an spa-3000 that works perfectly well now. (UK) I had some trouble at first though. What firmware are you using and what's the symptom? does it not hang up or does it hang up during calls? Conrad Thanks Conrad, It is brand new so I assume the firmware is the latest?: Software Version: 3.2.6(GWa) Hardware Version: 1.1.5. It just doesn't detect real hangups at all. If the person calling hangs up, either before and after the call is answered, the SPA will eventually timeout after about 30 seconds and hang up - in other words it does not detect the disconnect tone like it should. I have other small niggles but I'm sure I can sure them with some config tweeking but right now the hangup problem is really my top priority. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
It is brand new so I assume the firmware is the latest?: Software Version: 3.2.6(GWa) Hardware Version: 1.1.5. It just doesn't detect real hangups at all. If the person calling hangs up, either before and after the call is answered, the SPA will eventually timeout after about 30 seconds and hang up - in other words it does not detect the disconnect tone like it should. Interesting, I had it the other way round. It detected the disconnect tone during conversations. I disabled disconnect tone detection. *I think* it detects a polarity reversal instead. (It's been a while and once it worked I forgot about it) I posted my settings here http://www.conradwood.net/sipura.pdf Are they any different from yours? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE:Asterisk and dialer Running on Thin Clients
Vitaly, could you please be more spesific about all you did in order to get tis done, ill do anithing to aconplish this. Thank You! On 10/24/06, Vitaly Oborsky [EMAIL PROTECTED] wrote: SorryYou can, but it will demand a lot of work. We now work aboveintroduction of such decision on thin clients under control of thinstation. As softphone it is used mozphone (front-end), from thethin client network_client (back-end).___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax problem
Brian Candler wrote: On Mon, Oct 23, 2006 at 08:23:12PM -0700, Lee Howard wrote: If you don't mind saying, what is missing for full t.38 support? Steve giving Digium a royalty-free license to his GPL software or a pure-GPL branch of the Asterisk codebase, take your pick. Why royalty-free? AFAICS there's nothing to stop Digium licensing this code commercially from him, if it adds value to the product. You've misunderstood something. Digium will not commit anything to the Asterisk code base that is not disclaimed to them first. They do this for various commercial purposes. They *could* take anything GPL, like spandsp or any related T.38 developments in OpenPBX, and commit it to a GPL-only branch of the Asterisk codebase, but then they would have features missing from their non-GPL licensed commercial offering. So yes, there is nothing to stop Digium from using GPL code in their GPL Asterisk ... except Digium stopping themselves. And they do that rather predictably... http://bugs.digium.com/view.php?id=7742 ... for whatever purpose they may have in keeping hardware support and features from Asterisk. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic Codec Selection
Hi, Does anyone know a what to use a different codec for calls which are handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? The idea is to reduce the bandwidth to the server for the majority of calls, but get good quality on internal calls. With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail idea and a question
When you listen to old messages, it would be better if Asterisk reversed the order so that it starts at the most recent message and then forwarding goes to the next oldest message, etc... The last message would be the oldest. This makes more sense for old messages. Also, is there a way to have it so that after one message plays, the next one plays automatically without having to press 6? This would be very useful when checking your messages remotely say from a handsfree car phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UA - number assignment
I think I understood what you want:
1- You want when someone dials an extension, do a Lookup in a database
using FWDCIDNAME
2- Then Dial the number that corresponds to this FWDCIDNAME in database
is that?
If it is so, i would recomend you to use AstDB - Asterisk Berkeley DB
(version1) - automatically installed with your asterisk.
Example:
exten=_X.,1,Set(NumberToDial=DB(myuserlist/${FWDCIDNAME})
exten= _X.,2,Dial(SIP/${NumberToDial})
exten= _X.,3,hangup
Take a look on this function and applications on your CLI show function DB
hope it helps.
Pls give me some feedback
On 10/24/06, Paul Ianas [EMAIL PROTECTED] wrote:
My problem is simple and I've issued it about 3 weeks ago. I want the UAs to
authenticate with a number to the SIP server. Is this possible?
For example, I configured an AT-RG613TX (Allied Telesyn Residential
Gateway). In its configuration it is not possible for me to skip specifying
a number (ex. 102) along with the username. I've looked into the source code
(SIP implementation) of Asterisk and, as I figured out, it is not possible
to tell Asterisk the number the user has.
The question is: how can I assign a number to a user in Asterisk? One
solution would be to define two rules in extensions.conf :
exten = 102,1,SetCallerId,${FWDCIDNAME}
exten = 102,2,Dial(SIP/pianas)
these would tell Asterisk that user pianas has the number 102.
Is there any other solution for my problem? (a database for example).
Thank you.
--
Paul Ianas
Programming Engineer
Level 7 Software
Timisoara, 59D Bucovinei
phone: 0744137020
email: [EMAIL PROTECTED]
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Re: [asterisk-users] Becoming a User on IRC
You cant PM anyone if you arent registerd. When you message nickserv copy exaclty how it is written in the MOTD (except the password part). - Original Message - From: Eddie Johnson Jr To: asterisk-users@lists.digium.com Sent: Tuesday, October 24, 2006 2:13 PM Subject: [asterisk-users] Becoming a User on IRC Hello, I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? Thanks, Ed ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Conrad Wood wrote: It is brand new so I assume the firmware is the latest?: Software Version: 3.2.6(GWa) Hardware Version: 1.1.5. It just doesn't detect real hangups at all. If the person calling hangs up, either before and after the call is answered, the SPA will eventually timeout after about 30 seconds and hang up - in other words it does not detect the disconnect tone like it should. Interesting, I had it the other way round. It detected the disconnect tone during conversations. I disabled disconnect tone detection. *I think* it detects a polarity reversal instead. (It's been a while and once it worked I forgot about it) I posted my settings here http://www.conradwood.net/sipura.pdf Are they any different from yours? Conrad There are some differences, yes. Before I begin I should say that I think I've sorted out my main problem. I basically experimented with putting the values that were in the disconnect detect field into the ring tone field so I could hear them. Although the 400Hz tone was correct with my initial setting [EMAIL PROTECTED];20(*/0/1) and subsequent settings (including the same as you have), they just didn't sound right. There would be two distinct volumes on the 400Hz tone. In the end I'm using [EMAIL PROTECTED];2(0/*/1). This may seem backwards but seems to work. The higher the value of the number of repeats (2 in my case), the longer it takes to detect the disconnect tone. Also in this version of the firmware it does not seem to be necessary to put two tone values. But yes we have differences in our config other than that. You have polarity reversal detection and I do not (I did try with it on, but it didn't help even though there I have measured a polarity reversal on disconnect) You also have long silence detection off, which would seem logical to me. I have it on, but will probably switch it off in case it drops the line if I put the phone down to look for something etc. You have min CPC at 0.085 and mine is at 0.09. We also differ slightly at the bottom of the page. You have 3ms On hook speed, I have less than 5ms. You have Line In Use Voltage 30 and I have 25. You have Ring Validation 100Ms and I have 256. You have Ring Indication Delay of 256 and I have 0. I will now try your settings to see if it helps with my next big problem --- I'm not getting a CLI number. Instead I get the Username I've allocated to my SPA. Once again thanks hugely for your help on this. It is really good to be able to compare configs. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
You have polarity reversal detection and I do not (I did try with it on, but it didn't help even though there I have measured a polarity reversal on disconnect) FWIW: I once had a nasty DSL filter that broke polarity reversal detection. You have 3ms On hook speed, I have less than 5ms. You have Line In Use Voltage 30 and I have 25. You have Ring Validation 100Ms and I have 256. You have Ring Indication Delay of 256 and I have 0. I had problems with my (old) phone ringing briefly at some stage, so I experimented a little. I will now try your settings to see if it helps with my next big problem --- I'm not getting a CLI number. Instead I get the Username I've allocated to my SPA. ah. Do you have callerid from BT (bt line?). I signed up for something called BT Privacy or so which is free and gives you callerid. If you turn on logging (debug) on the sipura it'll log the received callerid via syslog. Also helpful to check under info Last seen number or so. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Just a thought ... try reversing the Tip and Ring Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You have polarity reversal detection and I do not (I did try with it on, but it didn't help even though there I have measured a polarity reversal on disconnect) FWIW: I once had a nasty DSL filter that broke polarity reversal detection. You have 3ms On hook speed, I have less than 5ms. You have Line In Use Voltage 30 and I have 25. You have Ring Validation 100Ms and I have 256. You have Ring Indication Delay of 256 and I have 0. I had problems with my (old) phone ringing briefly at some stage, so I experimented a little. I will now try your settings to see if it helps with my next big problem --- I'm not getting a CLI number. Instead I get the Username I've allocated to my SPA. ah. Do you have callerid from BT (bt line?). I signed up for something called BT Privacy or so which is free and gives you callerid. If you turn on logging (debug) on the sipura it'll log the received callerid via syslog. Also helpful to check under info Last seen number or so. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a User on IRC
On Tue, Oct 24, 2006 at 08:13:04AM -0400, Eddie Johnson Jr wrote: I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username This is a strange way to attempt to write to the channel Asterisk itself. You nee dot be registered first. And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? See http://freenode.net/faq.shtml#nicksetup -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
ah. Do you have callerid from BT (bt line?). I signed up for something called BT Privacy or so which is free and gives you callerid. If you turn on logging (debug) on the sipura it'll log the received callerid via syslog. Also helpful to check under info Last seen number or so. There is CLI on this particular line. I even managed to get it to work with the TD400P (or whatever the analog card with 4 modules is that Digium sells for Asterisk) in the past -- but no luck with hangup detection on that at all so I gave up on it. I'm not seeing any caller id in the syslog nor the last seen number thing. (which helpfully just says , :-) I thought your suggestion about the filter was excellent so I tried a few different ones (we are an ISP so I have a few hanging around ;-) ) but to no avail. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio half way through call
I am getting the following on my server when the problem happens: Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-209 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-210 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-211 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-211 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-211 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 208 not within window 209-212 Any idea what this means? To me it looks like it just is missing a packet, but why does it not continue? On 10/23/06, Matt [EMAIL PROTECTED] wrote: Have you tried disabling the jitterbuffer? Maybe it is a bug in the jitterbuffer code, then? On 10/23/06, Pavel Jezek [EMAIL PROTECTED] wrote: I have same problem, but with 1.4 branch, after several minutes, asterisk stops sending packets resulting one way audio, this problem appears especialy when bigger jitter appears (300ms) on one connection (I have jitterbuffer enabled on IAX), bigger jitter resulting in bigger one way audio probability in my case... PJ Matt wrote: Hi, I have asterisk 1.2.12 running on my server. Everything seems to be working fine on it. It has an IAX connection to the terminator/orignator. Again, everything seems to be fine.. calls come in and go out. However, it seems that after a call has been up for several minutes audio will go one-way. That is, we can hear the other person, but they can not hear us. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Henry.L.Coleman wrote: Just a thought ... try reversing the Tip and Ring Henry L.Coleman CEO Henry, Apologies for answering the wrong message in my last post. I thought I was answering the one from Conrad. Sorry! By reversing the Tip and Ring you mean physically in the wiring or somewhere in the SPA? I can see Forward/Reverse settings for Line1 in the config, but nothing on the PSTN side? Thanks, Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio half way through call
Just as a follow up.. on the OTHER server that is connected I'm seeing: chan_iax2.c: Received VNAK: resending outstanding frames On 10/24/06, Matt [EMAIL PROTECTED] wrote: I am getting the following on my server when the problem happens: Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-209 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-210 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-211 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-211 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not within window 209-211 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 208 not within window 209-212 Any idea what this means? To me it looks like it just is missing a packet, but why does it not continue? On 10/23/06, Matt [EMAIL PROTECTED] wrote: Have you tried disabling the jitterbuffer? Maybe it is a bug in the jitterbuffer code, then? On 10/23/06, Pavel Jezek [EMAIL PROTECTED] wrote: I have same problem, but with 1.4 branch, after several minutes, asterisk stops sending packets resulting one way audio, this problem appears especialy when bigger jitter appears (300ms) on one connection (I have jitterbuffer enabled on IAX), bigger jitter resulting in bigger one way audio probability in my case... PJ Matt wrote: Hi, I have asterisk 1.2.12 running on my server. Everything seems to be working fine on it. It has an IAX connection to the terminator/orignator. Again, everything seems to be fine.. calls come in and go out. However, it seems that after a call has been up for several minutes audio will go one-way. That is, we can hear the other person, but they can not hear us. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distributing calls among channels in dial group
Hi everybody! Is it possible to order Asterisk to distribute calls to ZAP channels belonging to one channel group (also called dial group) in any other way than in sequential order (1,2,3 etc.)? I would like to distribute calls equally between all available PRI spans. Thanks in advance for any tip! Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] INVAL Messages
Hi Marian, I think we worked it out... (time will tell now).. Our gateway people were able to put IAX2 debug on, and then filter the trace (manually!) so that we could compare call-flow. Heres what they saw: lon-pbx-backup-1*CLI Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 6ms SCall: 00074 DCall: 0 [xx.xx.xx.xx:4569] lon-pbx-backup-1*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 6ms SCall: 1 DCall: 00074 [xx.xx.xx.xx:49308] lon-pbx-backup-1*CLI Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 9ms SCall: 1 DCall: 00074 [xx.xx.xx.xx:49308] AUTHMETHODS : 3 CHALLENGE : 552508132 USERNAME: ubigradin lon-pbx-backup-1*CLI Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 08017ms SCall: 00074 DCall: 0 [xx.xx.xx.xx:4569] CAUSE CODE : 0 lon-pbx-backup-1*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 0 DCall: 00074 [xx.xx.xx.xx:49308] We don't quite understand is the HANGUP.. but we did note the 49308 port.. So the NAT router was changing to different ports. We believe that a Draytek router uses two definitions for NATing... Port Forwarding and Open Ports. We've switched to Open Ports (which really seems like its intended for a RANGE of ports, but we've specified only one. Now when we ethereal our WAN connection, we see 4569 in both directions.. Hopefully solved... Thanks! Adrian Marsh Tel: +44 (0) 20 71833427 Fax: +44 (0) 1793 441594 http://www.ubiquisys.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marian Rychtecky Sent: 24 October 2006 11:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] INVAL Messages Hi Adrian, yes this problem has happen only sometime, but i dont know exactly when i has discover this - plase read my comments: A)Calling directly via public IP's (port 4569 is forwarded on ADSL modem to asterisk1) - not working Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 4 DCall: 0 [213.160.177.186:4569] - here is source port of transmiting packet 4569 (my site) VERSION : 2 CALLED NUMBER : 1299 CODEC_PREFS : () CALLING NUMBER : 1199 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Marian_Rychtecky LANGUAGE: en USERNAME: some_username FORMAT : 2 CAPABILITY : 2097151 ADSICPE : 2 DATE TIME : 2006-10-18 10:16:14 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 6ms SCall: 3 DCall: 4 [213.160.177.186:9785] -- but here i got response from port 9785 (other site, because of NAT translation on other site) AUTHMETHODS : 3 CHALLENGE : 585590037 USERNAME: VALSABBIA-SLOVENSKO Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 3 [213.160.177.186:9785] B) calling thru openvpn - working Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 4ms SCall: 1 DCall: 0 [192.168.255.2:4569] -- here im sending the same packet to port 4569... VERSION : 2 CALLED NUMBER : 1299 CODEC_PREFS : () CALLING NUMBER : 1199 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Marian_Rychtecky LANGUAGE: en USERNAME: user_name FORMAT : 2 CAPABILITY : 2097151 ADSICPE : 2 DATE TIME : 2006-10-18 10:14:16 -- Called VALSABBIA-SLOVENSKO:[EMAIL PROTECTED]/1299 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00012ms SCall: 1 DCall: 1 [192.168.255.2:4569] --- here i got reply from port 4569 (and this port is excepted) AUTHMETHODS : 3 CHALLENGE : 186694617 USERNAME: VALSABBIA-SLOVENSKO Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00034ms SCall: 1 DCall: 1 [192.168.255.2:4569] MD5 RESULT : b0674601456416db7e474de9a858c742 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT so get ACCEPT and evething is working Timestamp: 00041ms SCall: 1 DCall: 1 [192.168.255.2:4569] FORMAT : 2 So i suppose that is the problem with comparing of source and destination port of IAX2 packets Look at your tcpdump and write me plase if you have the same troubles. Good luck,
[asterisk-users] Resampling Audio for use with Asterisk
Hello All,I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all generate files that sound like a bad acid trip. I tried increasing the speed 551.25 percent after doing the resample on these files and then it sounds like something from the wizard of oz...Any help that you can provide would be appreciated.Thanks,-Nate ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
I'm not seeing any caller id in the syslog nor the last seen number thing. (which helpfully just says , :-) I'd be pretty sure that the device doesn't detect the cli. My one does list the number under the 'last seen number thing'. What sort of line is it? Straight BT? telewest? Some converter? I thought your suggestion about the filter was excellent so I tried a few different ones (we are an ISP so I have a few hanging around ;-) ) but to no avail. Thanks. It's unlikely that it would affect cli. It was meant as a possible explanation for the disappearing polarity reversal. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] misdn.conf: how to set music on hold
Hi, is there anybody who knows how to set music on hold for an ISDN channel? In zapata.conf there is musiconhold parameter. Is there something similar for misdn.conf? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributing calls among channels in dial group
Define diferent trunks for every PRI span and use RANDOM on your dialplan before dialing! On 10/24/06, Asterisk [EMAIL PROTECTED] wrote: Hi everybody! Is it possible to order Asterisk to distribute calls to ZAP channels belonging to one channel group (also called dial group) in any other way than in sequential order (1,2,3 etc.)? I would like to distribute calls equally between all available PRI spans. Thanks in advance for any tip! Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro 'exited non-zero'
Can someone tell me if this indicates a problem? What does it mean when a macro exits != 0 ? Spawn extension (macro-syst_FindAppServer, s, 5) exited non-zero on 'SIP/xxx.yyy.142.186-b7515f98' in macro 'syst_FindAppServer' Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Resampling Audio for use with Asterisk
Hi, I'm having no trouble using: sox yourinputfile.wav -r 8000 -c 1 youroutfile.al resample -ql Regards, Tristan Nate Criss a crit: Hello All, I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all generate files that sound like a bad acid trip. I tried increasing the speed 551.25 percent after doing the resample on these files and then it sounds like something from the wizard of oz... Any help that you can provide would be appreciated. Thanks, -Nate ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
We had/have this problem, too - we eventually got it working (just by constantly rebooting it), but it seems that something's not working properly somewhere.. Can you look in your phone's boot log and see if you are getting any errors? We were seeing errors relating to the phone not being able to read sip.ld properly. CP On 23-Oct-06, at 5:51 PM, Edwin Lam wrote: i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at Updating initial configuration... screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the setting works fine w/ SP501 with bootrom 3.1.2/sip 1.6.3 -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Audiocodes MP-20x
I will sign in with good experiences with MP124 and Mediant 1000. I have an MP202 under test. --On Tuesday, October 24, 2006 10:10 AM +0300 Paul Ianas [EMAIL PROTECTED] wrote: I have used AudioCodes MP 102, 104 and 108, both FXS and FXO. I have also used AudioCodes Mediant 2000. I can tell you that these are good devices. There are also many other media gateways that have a lot of facilities, but many of these implement those facilities in software. AudioCodes has also a quite good – let's say -- hardware support. I haven't used MP20x. -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone: 0744137020 email: [EMAIL PROTECTED] __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Monday, October 23, 2006 1:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Audiocodes MP-20x Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Conrad Wood wrote: I'm not seeing any caller id in the syslog nor the last seen number thing. (which helpfully just says , :-) I'd be pretty sure that the device doesn't detect the cli. My one does list the number under the 'last seen number thing'. What sort of line is it? Straight BT? telewest? Some converter? I thought your suggestion about the filter was excellent so I tried a few different ones (we are an ISP so I have a few hanging around ;-) ) but to no avail. Thanks. It's unlikely that it would affect cli. It was meant as a possible explanation for the disappearing polarity reversal. Conrad I think I've found where the polarity reversal is going ... I think my lightning/spike filter is eating it or something. When I look at the syslog with the filter removed I see messages about polarity reversals. With the filter they are missing. Yet the phones I normally have plugged in still seem to read the CLI with no difficulty with or without the filter, and the SPA can't read them with or without the filter. Very fustrating. Yes, it is a bog standard BT line. I've tried using the CLI detection mode without the PR but that doesn't work either. I'm not sure what to try next Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK
Yep, just swop the two wires. Sometimes the Tip and Ring get reversed and most loop start interfaces don't really care (they work either way). It's worth a try since if the disconnect is a reverse polarity flash then the card may see not see this condition as it is already reversed. I have a similar problem with Foriegn Exchange line (FX) but I haven't had time to visit the client to check this out yet. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Just a thought ... try reversing the Tip and Ring Henry L.Coleman CEO Henry, Apologies for answering the wrong message in my last post. I thought I was answering the one from Conrad. Sorry! By reversing the Tip and Ring you mean physically in the wiring or somewhere in the SPA? I can see Forward/Reverse settings for Line1 in the config, but nothing on the PSTN side? Thanks, Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE:Asterisk and dialer Running on Thin Clients
Ignacio Ortega A. a écrit : *Vitaly,* could you please be more spesific about all you did in order to get tis done, ill do anithing to aconplish this. Have a look at the mailing list archive of MozPhone (moziax.mozdev.org): back in August, Machula Viach made modifications in order to run MozPhone in TSE environment. I plan to integrate his changes but it's not done yet. MozPhone was specifically developped with thin client in mind (complete separation of IAX / sound processing running on the client and user interface running on the server); and I have successfully used it with LTSP (www.ltsp.org), though not for hundreds of clients. Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro 'exited non-zero'
It means that it exited at priority 5 of the s extension in that context. (.. s, 5)It does not inherently mean anything bad, depending on if that is an accurate exit point in your Macro.Anthony On 10/24/06, Douglas Garstang [EMAIL PROTECTED] wrote: Can someone tell me if this indicates a problem? What does it mean when a macro exits != 0 ?Spawn extension (macro-syst_FindAppServer, s, 5) exited non-zero on 'SIP/xxx.yyy.142.186-b7515f98' in macro 'syst_FindAppServer' Thanks,Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Anthony D Cennami ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] txfax only getting 1 page of 3.
Steve and everyone...
I am using spandsp snapshot from oct 12, 2006. I am using asterisk 1.2.13.
When I am sending faxes I am only getting partial pages.
Internally using an IAXY connected to the fax machine I get 1 page of 3.
Extenally to a fax service using TDM2401E card I get the same results.
Only partial pages for my fax. 1 or 2 pages of my 3 page fax.
exten = smvoice_faxout,1,txfax(${SMFAXFILE},caller,debug)
This is my fax line and I am not getting any debug information that I see.
Why am I only getting partial pages? My machine is 97% idle.
Thanks,
Jerry
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[asterisk-users] need help using tftp for polycom 501
Hi, I have a Polycom 501 that is currently unusable because I started a firmware and sip upgrade that I can't complete. The Ubuntu box address is set static at: 192.168.1.101. The phone address is set static at 192.168.1.51. The phone settings for the server menu are: Server Type: Trivial FTP Server Address: 192.168.1.101 Server User: PlcmSpIp Server Password: PlcmSpIp (not sure what it should be) Pro. Method: default I am using tcpdump to watch the network messages, and I see the phone sending messages like: 11:04:50.147597 IP 192.168.1.51.1025 192.168.1.101.69: 19 RRQ bootrom.ld octet 11:04:58.235875 IP 192.168.1.51.1027 192.168.1.101.69: 25 RRQ 0004f21136a1.cfg octet 11:06:36.728815 IP 192.168.1.51.1029 192.168.1.101.69: 25 RRQ .cfg octet I have the following files in the directory /srv/tftp: 0004f21136a1.cfg bootrom.ld phone774110.cfg sip.cfg I have edited 0004f21136a1.cfg to point to phone774110.cfg I get the following message on the phone: Could not contact boot server. error loading 004f21136a1.cfg If I ps -e I see tftp is active. I am at a total lose how to setup and use tftp properly. I have searched the Internet and read man pages, but I can't get it into my head. Any help will be very much appreciated. -- Marlin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi all, the lists seems to be littered with disconnect problems using various equipment (TDM 400,Linksys etc etc.) My question is very simple and could make for good solution to Asterisk users. Since * can detect various tones according to different country standards would it be possible to disconnect on the 'off-hook' warning tone? This tone is: 1400 Hz, 2060 Hz, 2450 Hz, and 2600 Hz, at a cadence of 0.1s on, 0.1s off. is it very easy to establish if this tone is present on the line simply ask the non-asterisk end to hangup and wait on the line if you hear a loud warning tone then that is the disconnect tone!. If this tone could be detected and issued as the # then * would see this as a dialled digit and force a disconnect. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail help
Title: Voicemail help I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know what would be the best Distro of Linux to use and version, what version of Asterisk works best to interact with CallManager, and what H323 ChannelType works. As you probably read in another thread I tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package). This doesn't seem to work to well, as somewhere along the line either CCM or OOH323 is disconnecting the call as soon as the playback application is run. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Becoming a User on IRC
Hello Dovid, My firsts time doing this what is MOTD? I also tried what you suggested /msg #asterisk username register and it did not work. I must not be doing something correct because I had a couple of other people try and not successful. Any suggetions? Ed From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dovid B Sent: Tuesday, October 24, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a User on IRC You cant PM anyone if you arent registerd. When you message nickserv copy exaclty how it is written in the MOTD (except the password part). - Original Message - From: Eddie Johnson Jr To: asterisk-users@lists.digium.com Sent: Tuesday, October 24, 2006 2:13 PM Subject: [asterisk-users] Becoming a User on IRC Hello, I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? Thanks, Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail help
I use Fedora Core and it works fine. I'm not connected to call manager though. which version of Asterisk are you using? -- Original message -- From: "Ward, Bill" [EMAIL PROTECTED] I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know what would be the best Distro of Linux to use and version, what version of Asterisk works best to interact with CallManager, and what H323 ChannelType works. As you probably read in another thread I tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package). This doesn't seem to work to well, as somewhere along the line either CCM or OOH323 is disconnecting the call as soon as the playback application is run. ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Voicemail help
I have tried both FC5 and 6. Asterisk works fine in both instances, for example when i connect with an IAX2 softclient like Idefisk. I only encounter the problem when I try to go through CCM. -Original Message- From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Tue 10/24/2006 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail help I use Fedora Core and it works fine. I'm not connected to call manager though. which version of Asterisk are you using? -- Original message -- From: Ward, Bill [EMAIL PROTECTED] I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know what would be the best Distro of Linux to use and version, what version of Asterisk works best to interact with CallManager, and what H323 ChannelType works. As you probably read in another thread I tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package). This doesn't seem to work to well, as somewhere along the line either CCM or OOH323 is disconnecting the call as soon as the playback application is run. winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
Carla Schroder wrote: On Monday 23 October 2006 17:38, Edwin Lam wrote: Re: [asterisk-users] Polycom SP4000 ftp problem From: Edwin Lam [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Carla Schroder wrote: Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't really an FTP server; it uses a different protocol, and uses only a single port (UDP 69). You can't use real FTP servers for this. sure if there's a tftp server that can provide the security and flexibility of ftp server. The difference between tftp and 'real' FTP servers is it does not ask for a login- that's why it's used for diskless clients and PXE net installs. ProFTP (and all other FTP servers) require a login authorization. This is usually invisible to the end-user on public FTP servers, but it's still there. So I'd look for how the phone authorizes itself to the ftp server. the authorization works fine. here's the log from proftpd: 10.1.3.54 UNKNOWN nobody [23/Oct/2006:15:53:48 -0700] USER sp4001 331 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PASS (hidden) 230 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PWD 257 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PASV 227 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] TYPE I 200 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] SIZE bootrom.ld 213 - it always stops at the SIZE bootrom.ld mesage. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
Edwin Lam wrote: Carla Schroder wrote: On Monday 23 October 2006 17:38, Edwin Lam wrote: Re: [asterisk-users] Polycom SP4000 ftp problem From: Edwin Lam [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Carla Schroder wrote: Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't really an FTP server; it uses a different protocol, and uses only a single port (UDP 69). You can't use real FTP servers for this. sure if there's a tftp server that can provide the security and flexibility of ftp server. The difference between tftp and 'real' FTP servers is it does not ask for a login- that's why it's used for diskless clients and PXE net installs. ProFTP (and all other FTP servers) require a login authorization. This is usually invisible to the end-user on public FTP servers, but it's still there. So I'd look for how the phone authorizes itself to the ftp server. the authorization works fine. here's the log from proftpd: 10.1.3.54 UNKNOWN nobody [23/Oct/2006:15:53:48 -0700] USER sp4001 331 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PASS (hidden) 230 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PWD 257 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PASV 227 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] TYPE I 200 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] SIZE bootrom.ld 213 - it always stops at the SIZE bootrom.ld mesage. rename bootrom.ld to something else like bootrom.ld-disabled. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a User on IRC
Hi Eddie, Connect to irc.freenode.net, and then type this: /msg nickserv register password nickserv will tell you that your nick is now registered. Then type this: /j #asterisk Say hi to CunningPike when you get there. CP On 24-Oct-06, at 1:12 PM, Eddie Johnson Jr wrote: Hello Dovid, My firsts time doing this what is MOTD? I also tried what you suggested /msg #asterisk username register and it did not work. I must not be doing something correct because I had a couple of other people try and not successful. Any suggetions? Ed From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, October 24, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a User on IRC You cant PM anyone if you arent registerd. When you message nickserv copy exaclty how it is written in the MOTD (except the password part). - Original Message - From: Eddie Johnson Jr To: asterisk-users@lists.digium.com Sent: Tuesday, October 24, 2006 2:13 PM Subject: [asterisk-users] Becoming a User on IRC Hello, I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? Thanks, Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )
We have a problem where callerid works 50% of the time on both lines. What we are seeing in the logs is: Oct 23 02:44:00 WARNING[28207] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 23 05:09:25 NOTICE[28840] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 23 05:09:27 WARNING[28840] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 24 02:06:12 NOTICE[29812] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 24 02:06:14 WARNING[29812] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 24 04:36:05 NOTICE[30440] chan_zap.c: Got event 2 (Ring/Answered)... Oct 24 04:36:07 WARNING[30440] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 24 15:22:21 NOTICE[30963] chan_zap.c: Got event 2 (Ring/Answered)... Oct 24 15:22:23 WARNING[30963] chan_zap.c: CID timed out waiting for ring. Exiting simple switch zapata.conf [channels] signalling=fxs_ks switchtype=national rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes cidsignalling=v23 ; UK CallerID cidstart=polarity ; UK CallerID hidecallerid=no sendcalleridafter=2 ; Magic for UK callerid callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=128 rxgain=2.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no faxdetect=incomming progzone=uk signalling=fxs_ks callerid=asreceived language=en context=business channel = 3 signalling=fxs_ks callerid=asreceived language=en context=daytime-analog channel = 4 Any help would greatly be appreciated. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Becoming a User on IRC
Unless of course the nick your using is used already in which case you will have to change it with /nick newnick -Original Message- From: [EMAIL PROTECTED] on behalf of Anthony Rodgers Sent: Tue 10/24/2006 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a User on IRC Hi Eddie, Connect to irc.freenode.net, and then type this: /msg nickserv register password nickserv will tell you that your nick is now registered. Then type this: /j #asterisk Say hi to CunningPike when you get there. CP On 24-Oct-06, at 1:12 PM, Eddie Johnson Jr wrote: Hello Dovid, My firsts time doing this what is MOTD? I also tried what you suggested /msg #asterisk username register and it did not work. I must not be doing something correct because I had a couple of other people try and not successful. Any suggetions? Ed From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, October 24, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a User on IRC You cant PM anyone if you arent registerd. When you message nickserv copy exaclty how it is written in the MOTD (except the password part). - Original Message - From: Eddie Johnson Jr To: asterisk-users@lists.digium.com Sent: Tuesday, October 24, 2006 2:13 PM Subject: [asterisk-users] Becoming a User on IRC Hello, I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? Thanks, Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models predicated on the ability of Caller ID to be transmitted to the PSTN with a value that is not expected; call centers are an obvious example, where customer-support staff make outbound calls with a Caller ID that may be from one of many possible clients. More troubling is the possibility that Caller ID may be used to trick unsuspecting call recipients into certain actions or beliefs, and it is this concern that's currently creating a legislative threat I believe must be averted. ... Congress is currently considering legislation titled The Truth in Caller ID Act, which certainly sounds noble. Who doesn't want correct Caller ID when receiving a call? The truth is that this bill is redundant--the Wire Fraud Act already covers this issue, and adding more wording seems to be merely a re-statement of a certain circumstance or type of Wire Fraud. While the wording of this legislation does not effectively change the amount of power a prosecutor currently has, I believe it will certainly create confusion and fear in the technical and investment community because of the uncertainty it promotes. It's like saying, I want you to not break the speeding laws AND I want you to not go over the speed limit! A legal staff could spend a week--at $200 an hour--explaining that to a CEO, despite the consistency. === http://www.oreillynet.com/pub/a/etel/2006/10/18/solving-the-caller-id-problem.html Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Becoming a User on IRC
Anthony, Thanks :) Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Tuesday, October 24, 2006 2:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a User on IRC Hi Eddie, Connect to irc.freenode.net, and then type this: /msg nickserv register password nickserv will tell you that your nick is now registered. Then type this: /j #asterisk Say hi to CunningPike when you get there. CP On 24-Oct-06, at 1:12 PM, Eddie Johnson Jr wrote: Hello Dovid, My firsts time doing this what is MOTD? I also tried what you suggested /msg #asterisk username register and it did not work. I must not be doing something correct because I had a couple of other people try and not successful. Any suggetions? Ed From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, October 24, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a User on IRC You cant PM anyone if you arent registerd. When you message nickserv copy exaclty how it is written in the MOTD (except the password part). - Original Message - From: Eddie Johnson Jr To: asterisk-users@lists.digium.com Sent: Tuesday, October 24, 2006 2:13 PM Subject: [asterisk-users] Becoming a User on IRC Hello, I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? Thanks, Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
On Tue, Oct 24, 2006 at 02:57:38PM -0400, Jay R. Ashworth wrote: This seems like a piece members of this list would find interesting... Further down, he notes: The PSTN cannot turn on a dime and restrict ANI/CLID from many clients using whitelist filters. Caller ID manipulation is used too widely for completely legitimate purposes, and any firm providing interconnection will almost always ask for a removal of the ingress filter when sending calls to another carrier. I believe that a check-ahead database that is consulted before call completion at any/every border is unworkable as a matter of cost and willpower. with which I disagree. In the current regulatory environment, the only thing they really have handle on is calls which transit the PSTN, and there are *already* rules which restrict what CNID may be transmitted across the PSTN by a LEC or IXC. Given that framework, my personal viewpoint is that that's *exactly* the situation, and that since most switches have that code in them already, though sometimes they don't bother to enable it, that this shouldn't be nearly as big a deal as he says it is. All they should have to do is instrument their ISDN trunks to see which ones are having customer-provided CNID sent down them, and clean up their datafill before enabling the restriction code that's already there. It's all about they money, though: if *every* LEC and IXC taking direct digital drops doesn't all force it at the same time, there will be scads of carrier changes. So perhaps legislation -- or more properly, enforcement of the current rules -- is called for. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disconnect problems and off-hook warning tone
On 2006-10-24 10:32:09 -0700, Henry.L.Coleman [EMAIL PROTECTED] said: Hi all, the lists seems to be littered with disconnect problems using various equipment (TDM 400,Linksys etc etc.) My question is very simple and could make for good solution to Asterisk users. Since * can detect various tones according to different country standards would it be possible to disconnect on the 'off-hook' warning tone? This doesn't seem very helpful, unless I am misunderstanding. It takes quite a while for this warning to start. Generally, I think these problems are about a delayed hangup detection, but rarely in my (very limited) experience does it persist long enough to get to the off hook warning tone. This tone is: 1400 Hz, 2060 Hz, 2450 Hz, and 2600 Hz, at a cadence of 0.1s on, 0.1s off . is it very easy to establish if this tone is present on the line simply ask the non-asterisk end to hangup and wait on the line if you hear a lou d warning tone then that is the disconnect tone!. If this tone could be detected and issued as the # then * would see thi s as a dialled digit and force a disconnect. I don't understand this last bit either... This could be a last defense against off hook PSTN lines. My biggest issue of this kind is with my gateway (wellgate 3701a) which doesn't sense the hangup very quickly and leaves me quite a few empty voicemails... I had it set to do polarity reversal detection, which fixed the voicemail, but seemed to hangup on some calls (which is far worse). Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 Beta 3 H323 Video?
This is probably the last time for a while is it possible to develop a quick and simple solution for this problem Audio works well, routing between SIP and h323... fine, but video still not providing any signalling. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )
(sorry for the second post) We have a problem where callerid works 50% of the time on both lines. What we are seeing in the logs is: Oct 23 02:44:00 WARNING[28207] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 23 05:09:25 NOTICE[28840] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 23 05:09:27 WARNING[28840] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 24 02:06:12 NOTICE[29812] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 24 02:06:14 WARNING[29812] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 24 04:36:05 NOTICE[30440] chan_zap.c: Got event 2 (Ring/Answered)... Oct 24 04:36:07 WARNING[30440] chan_zap.c: CID timed out waiting for ring. Exiting simple switch Oct 24 15:22:21 NOTICE[30963] chan_zap.c: Got event 2 (Ring/Answered)... Oct 24 15:22:23 WARNING[30963] chan_zap.c: CID timed out waiting for ring. Exiting simple switch zapata.conf [channels] signalling=fxs_ks switchtype=national rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes cidsignalling=v23 ; UK CallerID cidstart=polarity ; UK CallerID hidecallerid=no sendcalleridafter=2 ; Magic for UK callerid callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=128 rxgain=2.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no faxdetect=incomming progzone=uk signalling=fxs_ks callerid=asreceived language=en context=business channel = 3 signalling=fxs_ks callerid=asreceived language=en context=daytime-analog channel = 4 Any help would greatly be appreciated. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
On Tue, 2006-10-24 at 15:12 -0400, Jay R. Ashworth wrote: On Tue, Oct 24, 2006 at 02:57:38PM -0400, Jay R. Ashworth wrote: This seems like a piece members of this list would find interesting... Further down, he notes: The PSTN cannot turn on a dime and restrict ANI/CLID from many clients using whitelist filters. Caller ID manipulation is used too widely for completely legitimate purposes, and any firm providing interconnection will almost always ask for a removal of the ingress filter when sending calls to another carrier. I believe that a check-ahead database that is consulted before call completion at any/every border is unworkable as a matter of cost and willpower. with which I disagree. In the current regulatory environment, the only thing they really have handle on is calls which transit the PSTN, and there are *already* rules which restrict what CNID may be transmitted across the PSTN by a LEC or IXC. What's going on Jay... Round 2? (Kidding). I always ponder the stupidity in this act. How exactly do they expect to enforce this, and how long/short will it be before I myself take the first strike and sue either Dell or IBM for not posting the caller ID information from their outsourced vendors abroad. I mean really... You expect me to believe your name is John and you're from Seattle ven you shpeek to me like vat is my problem vis my computer? Vell type in v v v dart dell dart com... Call me an ass, call me rude... Call me realistic. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is overwriting proxy Via Header
Hi, I having a problem with my asterisk, it is overwriting the Proxy Via header with its own ip address and answering to the Proxy with the modified header, so the Proxy is having problems to route the response. I've tried with different versions of asterisk and nothing is changing, and if I try in other Server all works perfect, the problem is related with this particular server running over Linux dit_rs_poa_mtz_gw1.local 2.6.18 #1 SMP PREEMPT Fri Sep 22 10:43:25 BRT 2006 i686 i686 i386 GNU/Linux The scenary is like this: IPPhone---Proxy1--Asterisk Invite sent by the IPPhone INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:22149;branch=z9hG4bK554e149351ab7a3b From: teste sip:[EMAIL PROTECTED];tag=d772c33c63ebf84c To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:22149 Supported: replaces Proxy-Authorization: Digest username=5551125, realm=200.X.X.136, algorithm=MD5, uri=sip:[EMAIL PROTECTED], qop=auth, nc=0001, cnonce=eed75407c0d78607, opaque=4c4f15e2744c43bb0790c60a78c00552, nonce=453778dd3e3c605897e1efdeb823fc53122bc50c, response=67cab99628290773609250e828628f14 Call-ID: [EMAIL PROTECTED] CSeq: 63837 INVITE User-Agent: Grandstream BT110 1.0.8.23 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 388 Invite sent by the Proxy to * INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Contact: sip:[EMAIL PROTECTED]:50386 CSeq: 63837 INVITE From: teste sip:[EMAIL PROTECTED]:5060;tag=d772c33c63ebf84c Proxy-Authorization: digest username=5551125, realm=200.X.X.136, nonce=453778dd3e3c605897e1efdeb823fc53122bc50c, cnonce=eed75407c0d78607, response=67cab99628290773609250e828628f14, uri=sip:[EMAIL PROTECTED], opaque=4c4f15e2744c43bb0790c60a78c00552, qop=auth, nc=0001, algorithm=MD5 To: sip:[EMAIL PROTECTED]:5060 Via: SIP/2.0/UDP 200.X.X.136:5060;branch=z9hG4bKbced0281e38aa078 Via: SIP/2.0/UDP 192.168.1.100:22149;branch=z9hG4bK554e149351ab7a3b;received=201.X.X.212; rport=50386 Record-Route: sip:200.X.X.136:5060 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: Grandstream BT110 1.0.8.23 Call-Id: [EMAIL PROTECTED] Max-Forwards: 70 Content-Length: 389 supported: replaces content-type: application/sdp Trying sent by Asterisk with via modified SIP/2.0 100 Trying v: SIP/2.0/UDP 200.X.X.131:5060;branch=z9hG4bKbced0281e38aa078;received=200.X.X.136 v: SIP/2.0/UDP 192.168.1.100:22149;branch=z9hG4bK554e149351ab7a3b;received=201.X.X.212; rport=50386 f: teste sip:[EMAIL PROTECTED]:5060;tag=d772c33c63ebf84c t: sip:[EMAIL PROTECTED]:5060 i: [EMAIL PROTECTED] CSeq: 63837 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY m: sip:[EMAIL PROTECTED] l: 0 Could someone please tell me why asterisk is replacing proxy ip address with its own ip address in the last one via header ?? How can I solve it ? Regards, Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 goes one way audio when lag gets bad
Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw notransfer=yes trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 I have tried with jitterbuffer=no, and then rather then one-way-audio I get high packet loss until the connection settles back down.Any ideas on other things I can try? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with setting outbound caller id when calling another asterisk
I have an asterisk box at a remote location (which I will call remote), which registers to my local asterisk box (I'll call that one local), and uses that to route calls to the outside world. The problem I am having is that the remote location needs to lie about it's callerid sometimes, however if I set a callerid that matches the extension of another peer that exists, local returns a 403 to remote. I can set the callerid to the did and it will work fine, or I can set the callerid to something random and it will work fine. What does * do with the proxy-authorization header, because it seems to be ignoring the username part... or maybe I need to go read some RFCs. Any help is greatly appreciated. Thanks, Chris Mazuc -- SIP read from REMOTE:1025: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username=1XX1205, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=45a347bc, response=934b409f19a0ebf28d1cf266db29f497, opaque= Date: Tue, 24 Oct 2006 20:26:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 240 v=0 o=root 2238 2239 IN IP4 REMOTE s=session c=IN IP4 REMOTE t=0 0 m=audio 15384 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (14 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to REMOTE : 5060 (NAT) Found user '1XX1200' Reliably Transmitting (NAT) to REMOTE:1025: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;received=REMOTE;rport=1025 From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4 To: sip:[EMAIL PROTECTED];tag=as1f40e0ec Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] update_header: Unable to find our position
Hi i got lots of this from the asterisk console what does this mean?format_wav.c:247 update_header: Unable to find our positionasterisk console:Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[6010]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[3684]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[6010]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[6010]: format_wav.c:247 update_header: Unable to find our position Oct 24 16:39:19 WARNING[4430]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our positionOct 24 16:39:19 WARNING[3684]: format_wav.c:247 update_header: Unable to find our position -- Regards,Mark Quitoriano, CCNAFan the flame...http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
Eric ManxPower Wieling wrote: rename bootrom.ld to something else like bootrom.ld-disabled. did that. it hung on sip.ld, rename sip.ld, it hung on phone1.cfg. seems like if the file is bigger than say 1k. it'll hang. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] attempting native bridge on TDM2400
Hello list, I am encountering a bit of a problem in working with incoming calls with a TDM2400 and * 1.2.4; when a call comes in, * will correctly detect the ringing, but will sometimes report multiple Attempting native bridge. What I do is basically that when a call comes in, I dial a different box through the same Zaptel interface and I get logs like this: [call comes in] -- Starting simple switch on 'Zap/10-1' Oct 24 16:55:46 NOTICE[8457]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Oct 24 16:55:47 NOTICE[8457]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/10-1, Zap/g1|90|t) in new stack -- Called g1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered Zap/10-1 -- Attempting native bridge of Zap/10-1 and Zap/1-1 -- Attempting native bridge of Zap/10-1 and Zap/1-1 -- Attempting native bridge of Zap/10-1 and Zap/1-1 What I don't get is why * has a need for multiple Attempting native bridge on multiple calls flowing through the same interface. What does this mean? Best regards l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASterisk Start problem
Hi all I have installed 1.2.12.1 in FC5 with libpri.1.2.4 when i start iam getting the following error and it quits == Registered channel type 'Local' (Local Proxy Channel Driver)[chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory Oct 23 16:16:07 WARNING[11084]: loader.c:554 load_modules: Loading module chan_zap.so failed![EMAIL PROTECTED] agc]# Ouch ... error while writing audio data: : Broken pipe what is the problem, any suggestions ? Ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASterisk Start problem
On Mon, Oct 23, 2006 at 04:17:45PM +0530, ram wrote: Hi all I have installed 1.2.12.1 in FC5 with libpri.1.2.4 when i start iam getting the following error and it quits == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory Oct 23 16:16:07 WARNING[11084]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] agc]# Ouch ... error while writing audio data: : Broken pipe what is the problem, any suggestions ? Is libpri installed? Where exactly? -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
J I have seen your paper on the caller ID issue I can't agree with you more. On 10/24/06, J. Oquendo [EMAIL PROTECTED] wrote: On Tue, 2006-10-24 at 15:12 -0400, Jay R. Ashworth wrote: On Tue, Oct 24, 2006 at 02:57:38PM -0400, Jay R. Ashworth wrote: This seems like a piece members of this list would find interesting... Further down, he notes: The PSTN cannot turn on a dime and restrict ANI/CLID from many clients using whitelist filters. Caller ID manipulation is used too widely for completely legitimate purposes, and any firm providing interconnection will almost always ask for a removal of the ingress filter when sending calls to another carrier. I believe that a check-ahead database that is consulted before call completion at any/every border is unworkable as a matter of cost and willpower. with which I disagree. In the current regulatory environment, the only thing they really have handle on is calls which transit the PSTN, and there are *already* rules which restrict what CNID may be transmitted across the PSTN by a LEC or IXC. What's going on Jay... Round 2? (Kidding). I always ponder the stupidity in this act. How exactly do they expect to enforce this, and how long/short will it be before I myself take the first strike and sue either Dell or IBM for not posting the caller ID information from their outsourced vendors abroad. I mean really... You expect me to believe your name is John and you're from Seattle ven you shpeek to me like vat is my problem vis my computer? Vell type in v v v dart dell dart com... Call me an ass, call me rude... Call me realistic. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASterisk Start problem
Did you compile and install these in the correct order: zaptel libpri asterisk CP On 23-Oct-06, at 5:47 AM, ram wrote: Hi all I have installed 1.2.12.1 in FC5 with libpri.1.2.4 when i start iam getting the following error and it quits == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory Oct 23 16:16:07 WARNING[11084]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] agc]# Ouch ... error while writing audio data: : Broken pipe what is the problem, any suggestions ? Ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme... No channel type registered for 'zap'
When I call meetme: exten = 1000,1,Answer exten = 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in new stack -- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, |||d) in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not heavily tested! Oct 24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No channel type registered for 'zap' Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '5000' -- Playing 'conf-onlyperson' (language 'en') -- Hungup 'IAX2/xxx.yyy.142.204:4569-2' However, I have zaptel and ztdummy drivers loaded: demeter:(acd1)asterisk # lsmod Module Size Used by ztdummy 3464 0 zaptel218756 1 ztdummy usbhid 31328 0 ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 siimage 9472 0 [permanent] piix8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 usbcore84740 5 usbhid,ohci_hcd,ehci_hcd,uhci_hcd rtc10164 1 ztdummy crc_ccitt 2176 1 zaptel and I have the app_meetme application loaded. *CLI show modules like meetme Module Description Use Count app_meetme.so MeetMe conference bridge 0 1 modules loaded What's up with that? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax problem
But if we have asterisk and add on Steve's code wouldn't it (suppor to recieve a t.38 fax call and have spandsp decode it) work? What does Steve granting a license to Digium have to do with it? I don't care if Asterisk and the fax support don't come from the same place. On 10/23/06, Lee Howard [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: On 10/23/06, *Steve Underwood* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The latest test versions of the spandsp support both ECM and T.38. The T.38 functionality may not be available with Asterisk, though. Is there any other software it would work with? Yes, OpenPBX. Is there any situation under which it might work?I don't really know how to respond to the ambiguity in that question. If you don't mind saying, what is missing for full t.38 support?Steve giving Digium a royalty-free license to his GPL software or apure-GPL branch of the Asterisk codebase, take your pick. Also with ECM being present now, that should eliminate distortion? ECM remedies data corruption and not image distortion caused by a brokenviewer. Its at random places during the fax there are glitches such as parts of the line missing or being shifted a bit Yes, ECM probably will address that.Lee.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update_header: Unable to find our position
On Wed, 2006-10-25 at 04:44 +0800, Mark Quitoriano wrote: Hi i got lots of this from the asterisk console what does this mean? format_wav.c:247 update_header: Unable to find our position asterisk console: Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position my first guess would simply be a wav file that's broken. You could try to re-encode it with sox and see if that fixes it. But really, that's more instinct than anything. Haven't looked at that code at all. conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail idea and a question
When you listen to old messages, it would be better if Asterisk reversed the order so that it starts at the most recent message and then forwarding goes to the next oldest message, etc... The last message would be the oldest. This makes more sense for old messages. Some people like it the way it is. It makes sense because you want to hear the sequence of messages. For example you will end up hearing (the way you want it). Msg 1. Hi I want 3l Msg 2 I am calling to order a camera. Wouldnt you want to know what they want first before you know how many ? Also, is there a way to have it so that after one message plays, the next one plays automatically without having to press 6? This would be very useful when checking your messages remotely say from a handsfree car phone. Pay some one on the dev list to code it in to asterisk so you can set an option in the dial plan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need help using tftp for polycom 501
Have you tried using just ftp ? - Original Message - From: Marlin Unruh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 24, 2006 7:28 PM Subject: [asterisk-users] need help using tftp for polycom 501 Hi, I have a Polycom 501 that is currently unusable because I started a firmware and sip upgrade that I can't complete. The Ubuntu box address is set static at: 192.168.1.101. The phone address is set static at 192.168.1.51. The phone settings for the server menu are: Server Type: Trivial FTP Server Address: 192.168.1.101 Server User: PlcmSpIp Server Password: PlcmSpIp (not sure what it should be) Pro. Method: default I am using tcpdump to watch the network messages, and I see the phone sending messages like: 11:04:50.147597 IP 192.168.1.51.1025 192.168.1.101.69: 19 RRQ bootrom.ld octet 11:04:58.235875 IP 192.168.1.51.1027 192.168.1.101.69: 25 RRQ 0004f21136a1.cfg octet 11:06:36.728815 IP 192.168.1.51.1029 192.168.1.101.69: 25 RRQ .cfg octet I have the following files in the directory /srv/tftp: 0004f21136a1.cfg bootrom.ld phone774110.cfg sip.cfg I have edited 0004f21136a1.cfg to point to phone774110.cfg I get the following message on the phone: Could not contact boot server. error loading 004f21136a1.cfg If I ps -e I see tftp is active. I am at a total lose how to setup and use tftp properly. I have searched the Internet and read man pages, but I can't get it into my head. Any help will be very much appreciated. -- Marlin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax problem
Andrew Joakimsen wrote: But if we have asterisk and add on Steve's code wouldn't it (suppor to recieve a t.38 fax call and have spandsp decode it) work? What does Steve granting a license to Digium have to do with it? I don't care if Asterisk and the fax support don't come from the same place. First off, I'm not really the right guy to be having this conversation, but since I know enough of the facts I can respond accurately enough to satisfy your query. If you ask a lot more questions in the what-if direction I may have to bow out. Steve's related code is two-fold... code that is in spandsp and code that is in OpenPBX. And, actually I think that spandsp comes with OpenPBX, so it's really just one download. Anyway, spandsp is a library. You get it, install it, and you end up with a bunch of code libraries that really don't do anything by themselves. You have to have some other software that utilizes those libraries, like the well-known txfax and rxfax applications... or like iaxmodem. Steve's work in OpenPBX is not really something that you can extract out of OpenPBX and stick into Asterisk very easily. I guess you're welcome to try, though. And, if you become successful in that - in producing a patch to apply onto Asterisk and you then endeavor to maintain that patch along with all of the other patches that you have to maintain to keep your motley Asterisk running your OpenVOX cards, your txfax/rxfax apps, and the myriad of other things that don't come with Asterisk for who-knows-what reason... well, then you're effectively maintaining your own little fork of Asterisk. And at that point I would wonder why you have gone through all of that effort just to avoid using OpenPBX, which is where Steve put that code for you to use in the first place. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic Conf
Hi there, I'm tring a basic asterisk settings. I have a asterisk 1.2.7.1 running on a I have a net with two computers and a router. The router IP in the local net is 192.168.1.1, The first pc has IP: 192.168.1.3 name datile3 . SO GNU Linux. the second pc has IP: 192.168.1.4 name fissun . SO GNU Linux. On datile3, it runs a softphone kphone. From this I want to call the external world. on fissun, it runs asterisk. On kphone I set as 192.168.1.4 as host and username as autentication username. when I run asterisk on fissun, I see only thos warning messages: Oct 25 01:31:55 WARNING[5549]: pbx.c:6438 ast_context_verify_includes: Context ' eutelia' tries includes nonexistent context 'out_eutelia' [...] Oct 25 01:31:55 WARNING[5549]: pbx.c:6438 ast_context_verify_includes: Context ' eutelia' tries includes nonexistent context 'out_eutelia' [...] Oct 25 01:31:55 WARNING[5549]: chan_iax2.c:9582 load_module: Unable to open IAX timing interface: No such file or directory If I run: sip show registry on asterisk (fissun) I see: HostUsername Refresh State voip.eutelia.it:5060 username8585 Registered It seems that all is all right. But, when I try to call a number from kphone (on datile3), I listen a message which say: the number you are dialing it does not exists. The .conf files: 1) extension.conf: [general] static=yes autofallthrough=yes clearglobalvars=no priorityjumping=no [eutelia] include = out_eutelia exten=_XX,1,Dial(SIP/[EMAIL PROTECTED],20) exten = _XX,2,Hangup 2) sip.conf: [general] context=eutelia realm=voip.eutelia.it port=5060 bindaddr=0.0.0.0 srvlookup=yes defaultexpirey=8600 useragent=Asterisk_Eut localnet=192.168.1.1/255.255.255.0 [out_eutelia] type=peer context=eutelia secret=xx username=username fromuser=username fromdomain=voip.eutelia.it host=voip.eutelia.it nat=yes dtmfmode=inband usereqphone=yes [datile3] type=friend host=dynamic username=datile3 context=eutelia permit=192.168.1.3 default=192.168.1.3 context=eutelia Is there anybody around who understand where is the problem? daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom provision errors still! Arg!
Cisco are worse. With the example files we were able to deploy and configure the Polycom phones with the newest firmware.With the sample files AND Cisco tech support we weren't even able to get them up to the latest version. On 10/23/06, Dean Collins [EMAIL PROTECTED] wrote: Lol, glad to hear it helped out that much.Yep polycoms are good but a real B**TCH to configure, I still do themmanually half the time.Cheers,Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] ] On Behalf Of Curt Shaffer Sent: Monday, 23 October 2006 10:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Polycom provision errors still! Arg! Shit I'll host him for free for that ;) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of DeanCollins Sent: Monday, October 23, 2006 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom provision errors still! Arg! No probs, maybe you should donate $5 to kerry's site to cover hosting fees? Cheers, Dean -Original Message- From: [EMAIL PROTECTED][mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Curt Shaffer Sent: Monday, 23 October 2006 9:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Polycom provision errors still! Arg! This absolutely helped. I downloaded those config files and copied then and change the name, addressing and such and it worked straight away! Must have been a munged config somehow! Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dean Collins Sent: Monday, October 23, 2006 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom provision errors still! Arg! Maybe this might help you. http://www.asterisktutorials.com/showproduct.php?ProductID=12 Cheers, Dean www.Mexuar.com -Original Message- From: [EMAIL PROTECTED] [mailto: asterisk-users- [EMAIL PROTECTED]] On Behalf Of Ivan Fetch Sent: Monday, 23 October 2006 7:31 PM To: Curt Shaffer Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom provision errors still! Arg! Hi, I believe he means to use the stock phone1.cfg and mac-address-of-the-phone.cfg files that come with the sip firmware you're running, and see if the phone will load those files. Ivan. On Mon, 23 Oct 2006, Curt Shaffer wrote: Do you mean .cfg and sip.cfg? Could you clarify for me please and Iwill try that. Thanks for the suggestion. Curt On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: What if you just use the default configuration files? On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from anFTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b_ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0/home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg|4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp|4|00|Initial log entry 1023201556|cdp|5|00|CDP is DISABLED. 1023201556|cdp|5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A
Re: [asterisk-users] Meetme... No channel type registered for 'zap'
Douglas Garstang wrote: When I call meetme: exten = 1000,1,Answer exten = 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in new stack -- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, |||d) in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not heavily tested! Oct 24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No channel type registered for 'zap' Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '5000' -- Playing 'conf-onlyperson' (language 'en') -- Hungup 'IAX2/xxx.yyy.142.204:4569-2' However, I have zaptel and ztdummy drivers loaded: demeter:(acd1)asterisk # lsmod Module Size Used by ztdummy 3464 0 zaptel218756 1 ztdummy usbhid 31328 0 ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 siimage 9472 0 [permanent] piix8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 usbcore84740 5 usbhid,ohci_hcd,ehci_hcd,uhci_hcd rtc10164 1 ztdummy crc_ccitt 2176 1 zaptel and I have the app_meetme application loaded. *CLI show modules like meetme Module Description Use Count app_meetme.so MeetMe conference bridge 0 1 modules loaded What's up with that? Doug. Doug, load chan_zap.so -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Meetme... No channel type registered for 'zap'
Doug wrote: When I call meetme: exten = 1000,1,Answer exten = 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in new stack -- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, |||d) in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not heavily tested! Oct 24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No channel type registered for 'zap' Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '5000' -- Playing 'conf-onlyperson' (language 'en') -- Hungup 'IAX2/xxx.yyy.142.204:4569-2' However, I have zaptel and ztdummy drivers loaded: demeter:(acd1)asterisk # lsmod Module Size Used by ztdummy 3464 0 zaptel218756 1 ztdummy usbhid 31328 0 ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 siimage 9472 0 [permanent] piix8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 usbcore84740 5 usbhid,ohci_hcd,ehci_hcd,uhci_hcd rtc10164 1 ztdummy crc_ccitt 2176 1 zaptel and I have the app_meetme application loaded. *CLI show modules like meetme Module Description Use Count app_meetme.so MeetMe conference bridge 0 1 modules loaded What's up with that? You don't happen to have a noload = chan_zap.so in /etc/asterisk/modules.conf, do you? What version of Asterisk and what channeltypes are loaded? Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall Installation
Hi Tzafrir, Thanks for your quick reply, I will look some downloads and install it as per your suggestion. I am using CentOS 4.3, kernel-2.6.9-34.01.EL Thanks again. Angel - Original Message From: Tzafrir Cohen [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, October 23, 2006 5:43:12 PMSubject: Re: [asterisk-users] Unicall Installation On Mon, Oct 23, 2006 at 02:11:22AM -0700, Angel Heart wrote: Hi, Thank you for your comment; Below was the result of./configure checking how to run the C++ preprocessor... /lib/cpp configure: error: C++ preprocessor "/lib/cpp" fails sanity check See `config.log' for more details. [EMAIL PROTECTED] libsupertone-0.0.2]# You don't have g++/gcc-c++ installed. You just need to install somepackages.Which Linux distribution do you use?-- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafriricq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406jabber:[EMAIL PROTECTED] http://www.xorcom.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic Conf
Hi there, I'm tring a basic asterisk settings. I have a asterisk 1.2.7.1 running on a I have a net with two computers and a router. The router IP in the local net is 192.168.1.1, The first pc has IP: 192.168.1.3 name datile3 . SO GNU Linux. the second pc has IP: 192.168.1.4 name fissun . SO GNU Linux. On datile3, it runs a softphone kphone. From this I want to call the external world. on fissun, it runs asterisk. On kphone I set as 192.168.1.4 as host and username as autentication username. when I run asterisk on fissun, I see only thos warning messages: Oct 25 01:31:55 WARNING[5549]: pbx.c:6438 ast_context_verify_includes: Context ' eutelia' tries includes nonexistent context 'out_eutelia' [...] Oct 25 01:31:55 WARNING[5549]: pbx.c:6438 ast_context_verify_includes: Context ' eutelia' tries includes nonexistent context 'out_eutelia' [...] Oct 25 01:31:55 WARNING[5549]: chan_iax2.c:9582 load_module: Unable to open IAX timing interface: No such file or directory If I run: sip show registry on asterisk (fissun) I see: HostUsername Refresh State voip.eutelia.it:5060 username8585 Registered It seems that all is all right. But, when I try to call a number from kphone (on datile3), I listen a message which say: the number you are dialing it does not exists. The .conf files: 1) extension.conf: [general] static=yes autofallthrough=yes clearglobalvars=no priorityjumping=no [eutelia] include = out_eutelia exten=_XX,1,Dial(SIP/[EMAIL PROTECTED],20) exten = _XX,2,Hangup 2) sip.conf: [general] context=eutelia realm=voip.eutelia.it port=5060 bindaddr=0.0.0.0 srvlookup=yes defaultexpirey=8600 useragent=Asterisk_Eut localnet=192.168.1.1/255.255.255.0 [out_eutelia] type=peer context=eutelia secret=xx username=username fromuser=username fromdomain=voip.eutelia.it host=voip.eutelia.it nat=yes dtmfmode=inband usereqphone=yes [datile3] type=friend host=dynamic username=datile3 context=eutelia permit=192.168.1.3 default=192.168.1.3 context=eutelia Is there anybody around who understand where is the problem? daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstFax Sending a Fax
Hi All I'm trying to understand how I would send my fax ? If I use Word or what ever word processor or even an email client to create what I want faxed. I have *asterisk setup with and FXO Gateway that will make the call to the fax number I dial SIP extension 320 is the FXO gateway. How do I now get my email or word document to TIFF to then fax to the FXO gateway or SIP/320 ? I don't understand that part. They all talk about an email with a TIF attachment and the TIF attachment is sent to the number in the subject line. Thanks all Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax problem
Andrew Joakimsen wrote: But if we have asterisk and add on Steve's code wouldn't it (suppor to recieve a t.38 fax call and have spandsp decode it) work? What does Steve granting a license to Digium have to do with it? I don't care if Asterisk and the fax support don't come from the same place. Its easy to maintain a well contained application, like rxfax, outside the tree. Trying to maintain patches to rtp.c, chan_sip.c and other core elements is too much of a pain to be reasonable. The code I contributed to Asterisk for T.38 passthrough languished for about 9 months before it was integrated. The day it was integrated it was in a less suitable state for integration than the day I contributed it. The patch to chan_sip.c had required many hours work from people over those 9 months, trying to keep up with the changes to the chan_sip.c in SVN. Its just wasteful and frustrating. The development of Asterisk has now degraded to the point where I will no longer contribute anything to it. If someone wants to take my code and make it work with Asterisk under GPL conditions, that's fine. The GPL gives you that right. Please make sure you stick to GPL conditions, though. You can't use G.729, for example, in an Asterisk that's using spandsp. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstFax Sending a Fax
You can use the fax server Hylafax ( http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem ) with IAXmodem ( http://iaxmodem.sourceforge.net/howto.php )You really don't want to be sending faxes over the internet via VoIP providers, not yet because there is no t.38 support for that. As long as the connection to the PSTN is on a card on the same machine or possibly over a network connection perhaps over a private line maybe using TDMoE then it should work fine On 10/24/06, Barry Fawthrop [EMAIL PROTECTED] wrote: Hi AllI'm trying to understand how I would send my fax ?If I useWordor what ever word processoror even an email client tocreate what I want faxed.I have *asterisk setup with and FXO Gateway that will make the call to the fax number I dialSIP extension 320is the FXO gateway.How do I now get my email or word document to TIFF to then fax to theFXO gateway or SIP/320 ?I don't understand that part.They all talk about an email with a TIF attachmentand the TIF attachment is sent to the number in the subject line.Thanks allBarry___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
